JPH0993166A - Loudspeaking information communication system - Google Patents

Loudspeaking information communication system

Info

Publication number
JPH0993166A
JPH0993166A JP24928495A JP24928495A JPH0993166A JP H0993166 A JPH0993166 A JP H0993166A JP 24928495 A JP24928495 A JP 24928495A JP 24928495 A JP24928495 A JP 24928495A JP H0993166 A JPH0993166 A JP H0993166A
Authority
JP
Japan
Prior art keywords
acoustic
impulse response
evaluation
signal input
reception signal
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
JP24928495A
Other languages
Japanese (ja)
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 JP24928495A priority Critical patent/JPH0993166A/en
Publication of JPH0993166A publication Critical patent/JPH0993166A/en
Pending legal-status Critical Current

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  • Filters That Use Time-Delay Elements (AREA)
  • Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)

Abstract

PROBLEM TO BE SOLVED: To suppress incidence probability of howling by inserting a line loss suitable for an impulse response characteristic so as to eliminate a reverberation component. SOLUTION: An acoustic evaluation signal caused by an evaluation signal generator 12 is sent under the operating environment by a reception signal output terminal 2. The acoustic evaluation signal receives the effect of an acoustic characteristic under external environment and inputted to the inside of the system from a transmission signal input terminal 4. The received acoustic replay signal is subjected to convolution arithmetic operation processing with a waveform of inverse characteristic to the acoustic evaluation signal by a high speed product sum arithmetic circuit 13. The impulse response characteristic of the acoustic space is obtained by the arithmetic processing and an impulse response characteristic evaluation device 14 calculates a square power change quantity in the acoustic impulse response and the quantity of the insertion loss of the circuit loss controller 15 is decided depending on the quantity of the square power change for a period set in the inside of the impulse response characteristic evaluation device 14.

Description

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

【0001】[0001]

【発明の属する技術分野】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
出現する音響反響成分を除去する音響反響除去装置を用
いた拡声情報通信システムに関するものである。
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 in which a signal of a receiving path appears in a transmitting path through an acoustic echo path. The present invention relates to a loudspeaker information communication system using an acoustic echo canceller that removes components.

【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.
It can be roughly divided into one that removes reflection caused by impedance mismatch of the line-to-four-line converter and one that removes reverberation due to acoustic coupling of speaker's voice in a loudspeaker telephone such as a video conference system. It is composed of a variable coefficient filter and a subtraction circuit that generate pseudo-acoustic echo. The basic operation of the acoustic reverberation removing device will be described below.

【0003】図6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
−カとマイクロホンへと接続されている。
FIG. 6 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 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. 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 / 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 = 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). Here, h = [h 1 , h 2 ,..., H n ] T (2) x = [x k−1 ,..., X kn ] T T : 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 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(196
8))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−キャンセラの室内
音場における適応性能の改善について”、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
Assuming s k , the estimated value ys k of y k is given by ys k = hs k T 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 (196
8)) 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, the e k = y k -ys k, is 0 <α ≦ 1 (5) e k is referred to as 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 according to the indoor characteristics (for example, Shoji Makino, Norio Koizumi: "Adaptive performance of the Eco-Canceller in the indoor sound field" On the improvement of ", IEICE (A), J71-A, 12, pp.2212-2214 (1988-12)).

【0006】[0006]

【発明が解決しようとする課題】学習同定法等の最小二
乗推定方式を基にした適応動作アルゴリズムは、入力信
号が白色雑音の時に最大の推定性能を発揮する事が出来
る。しかし、実際に入力とされる信号はスペクトラムに
偏りを持つ音声信号である為、対象システムここでは音
響特性を推定しきれず、残留音響反響が該送話信号内に
混入してしまい、通信音質を劣化させるだけでなく、最
悪の場合通信回線のル−プ利得が、0dBを越えてしま
う状況に陥り、ハウリングが発生する確率が極めて高く
なる。その対策として施される回線損失を挿入する事も
環境に対応できない為、逆に回線に大きなレベル変動を
発生させてしまう。従って、動作状態は不安定そのもの
で、高品質な音声情報通信空間を提供できない。
The adaptive operation algorithm based on the least-squares estimation method such as the learning identification method can exert the maximum estimation performance when the input signal is white noise. However, since the signal that is actually input is a voice signal that has a bias in the spectrum, the target system cannot estimate the acoustic characteristics here, and residual acoustic echo is mixed in the transmitted signal, which improves the communication quality. In addition to the deterioration, in the worst case, the loop gain of the communication line exceeds 0 dB, and the howling probability becomes extremely high. Inserting the line loss, which is applied as a countermeasure, cannot cope with the environment, and on the contrary, causes a large level fluctuation in the line. Therefore, the operating state itself is unstable, and a high quality voice information communication space cannot be provided.

【0007】又、本装置の様な拡声通信システムはさま
ざまな環境下で使用される事が予想され、その環境への
対応を確実に行なうのは困難であった。
In addition, it is expected that a loudspeaker communication system such as this device will be used in various environments, and it has been difficult to reliably handle such environments.

【0008】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用いた拡声
情報通信システムを提供する事を目的とする。
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.
It is an object of the present invention to provide a loudspeaker information communication system using an acoustic echo canceller that has high adaptive performance and performs acoustic control while always maintaining a large acoustic echo cancellation amount.

【0009】[0009]

【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
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を用いた拡声情報通信システムにお
いて、音響評価信号を生起する評価信号発生装置と、該
評価信号発生装置から発生された該音響評価信号が音響
空間を介して該送話信号入力端子より入力される応答信
号に畳み込み演算処理を加えて音響空間のインパルス応
答特性を算出する高速積和演算回路と、該インパルス応
答特性に適応した回線損失を挿入する事により残響成分
を除去することを特徴とする拡声情報通信システムを提
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. An acoustic echo canceller in which the coefficient sequence is sequentially updated by a coefficient correction amount calculation circuit is used so as to minimize the residual signal obtained by subtracting the pseudo acoustic echo generated by the variable coefficient digital filter from the echo component. In a loudspeaker information communication system, an evaluation signal generator for generating an acoustic evaluation signal, and the acoustic evaluation signal generated by the evaluation signal generator are transmitted via an acoustic space. A reverberation component is removed by inserting a high-speed product-sum calculation circuit that calculates the impulse response characteristic of acoustic space by adding convolution calculation processing to the response signal input from the signal input terminal, and a line loss that is adapted to the impulse response characteristic. (EN) Provided is a loudspeaker information communication system.

【0010】また、該インパルス応答特性を周波数変換
する高速フ−リエ変換装置と、該高速フ−リエ変換装置
からの出力となる空間周波数伝達特性から平均空間損失
を算出する入出力評価装置と、該平均空間損失を制御値
として通信回線の入出力レベルを決定する事によりハウ
リング発生の確率を極めて低減させた事を特徴とする請
求項1記載の拡声情報通信システムを提供する。
Further, a high-speed Fourier transform device for frequency-converting the impulse response characteristic, and an input / output evaluation device for calculating an average spatial loss from the spatial frequency transfer characteristic which is an output from the high-speed Fourier transform device, A loudspeaker information communication system according to claim 1, wherein the probability of howling occurrence is extremely reduced by determining the input / output level of the communication line using the average space loss as a control value.

【0011】さらに、該インパルス応答特性と、該空間
周波数伝達特性とを映像情報として使用者または設置者
に供給することで該回線損失の大きさ等の制御値を調整
できることを可能とし、より高品質な音声情報通信を実
現した請求項1記載の拡声情報通信システムを提供す
る。
Further, by supplying the impulse response characteristic and the spatial frequency transfer characteristic as video information to a user or an installer, it becomes possible to adjust a control value such as the magnitude of the line loss, and the like. A loudspeaker information communication system according to claim 1, which realizes high-quality voice information communication.

【0012】[0012]

【発明の実施の形態】以下、本発明の実施例を図面にも
とづいて説明する。図1は本発明の第一の高品質拡声情
報通信システムの構成を示すブロック図である。図1に
示す様に、本発明は従来の受話信号入力端子1、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11と同一構成の装置に、評価信号
発生装置12、高速積和演算回路13、インパルス応答
特性評価装置14、そして、回路損失制御装置15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11を用いた拡声情報通信システムにおいて、該評価
信号発生装置12により生起された該音響評価信号は受
話信号出力端子2より該拡声情報通信システム使用環境
下へと放出される。該音響評価信号は該外部環境下での
音響特性の影響を受け、該送話信号入力端子4より該拡
声情報通信システム内部へと入力される。入力された音
響応答信号は該高速積和演算回路13において該音響評
価信号の逆特性を有する波形と畳み込み演算処理が施さ
れる。この演算処理により音響空間のインパルス応答特
性が求まり、該インパルス応答特性評価装置14では、
該音響インパルス応答の自乗電力変化量を算出し、該イ
ンパルス応答特性評価装置14内部に設定された区間で
の自乗電力変化量の大小により、該回路損失制御装置1
5の挿入損失の大きさを決定する高品質拡声情報通信シ
ステム。図2は本発明の第二の高品質拡声情報通信シス
テムの構成を示すブロック図である。図1に示す様に、
本発明は従来の受話信号入力端子1、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11と同一構成の装置に、評価信号発生装置1
2、高速積和演算回路13、高速フ−リエ変換装置1
6、入出力評価装置17、そして、入出力補償装置18
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11を用いた拡声情報通信システムにおい
て、該高速積和演算回路13の出力である音響インパル
ス応答特性を該高速フ−リエ変換装置16において、周
波数変換を行い直接空間周波数伝達特性を算出し、この
値を基に該入出力評価装置17で零レベルがらの距離の
平均値を求める。そして、該入出力補償装置18では、
その距離の平均値に対応した補償レベルを決定し、通信
回線に挿入する高品質拡声情報通信システム。又、第一
の高品質拡声情報通信システムと第二の高品質拡声情報
通信システムを組み合わせた構成を取る事も可能であ
る。この場合、残響成分の混入しない高品質な音声通信
とハウリング発生の確率を極めて低く抑圧した拡声情報
通信システムを構成する事が可能となる。そして、該イ
ンパルス応答特性と、該空間周波数伝達特性とを映像情
報として使用者又は設置者に供給し、各環境別の特殊な
設定を行なえる様に構成された高品質拡声情報通信シス
テム。この評価信号を用いた諸設定値の決定は通信回線
の接続・非接続状態でも可能であり、又、自動・手動を
問わずに行なう事が可能である。勿論、環境の測定を行
わない場合には、該高速積和演算回路13はその動作を
停止せしめ、送話信号をそのまま通過させる様に制御さ
れる。
Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing the configuration of a first high-quality loudspeaker information communication system of the present invention. As shown in FIG. 1, according to the present invention, the conventional reception signal input terminal 1, reception signal output terminal 2, variable coefficient digital filter 3, transmission signal input terminal 4, subtraction circuit 5, transmission signal output terminal 6, coefficient correction. A device having the same configuration as the acoustic echo canceling device 11 that employs the learning identification method as an adaptive algorithm configured by the quantity calculation circuit 7, the reception signal input register 8, the pseudo impulse response register 9, and the product-sum calculation circuit 10 is evaluated. The configuration is such that a signal generator 12, a high-speed product-sum calculation circuit 13, an impulse response characteristic evaluation device 14, and a circuit loss control device 15 are added. 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 pseudo impulse which receives the reception signal of the reception signal input terminal 1 The variable coefficient digital filter 3 having the response register 9 and the variable coefficient digital filter 3 from the acoustic echo component of the reception signal input from the reception signal output terminal 2 to the transmission signal input terminal 4 via the acoustic echo path. In a loudspeaker information communication system using an acoustic echo canceller 11 in which a coefficient sequence is sequentially updated by the coefficient correction amount calculation circuit 7 so as to minimize a residual signal obtained by subtracting the pseudo acoustic echo generated in The acoustic evaluation signal generated by the evaluation signal generator 12 is emitted from the reception signal output terminal 2 to the environment for using the voice information communication system. The acoustic evaluation signal is influenced by the acoustic characteristics in the external environment, and is input from the transmission signal input terminal 4 to the inside of the voice information communication system. The input acoustic response signal is subjected to convolution calculation processing with the waveform having the inverse characteristic of the acoustic evaluation signal in the high-speed product-sum calculation circuit 13. By this arithmetic processing, the impulse response characteristic of the acoustic space is obtained, and in the impulse response characteristic evaluation device 14,
The circuit power loss controller 1 calculates the squared power variation of the acoustic impulse response and determines the magnitude of the squared power variation in a section set inside the impulse response characteristic evaluation device 14.
A high-quality loudspeaker information communication system for determining the size of insertion loss of 5. FIG. 2 is a block diagram showing the configuration of the second high-quality loudspeaker information communication system of the present invention. As shown in Figure 1,
The present invention includes the conventional reception signal input terminal 1, reception signal output terminal 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, reception signal. The evaluation signal generating device 1 has the same configuration as the acoustic echo canceling device 11 that employs a learning identification method as an adaptive algorithm configured by an input register 8, a pseudo impulse response register 9, and a product-sum operation circuit 10.
2, high-speed product-sum operation circuit 13, high-speed Fourier transform device 1
6, input / output evaluation device 17, and input / output compensation device 18
Has been 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 a loudspeaker information communication system using an acoustic echo canceling device 11 in which a coefficient sequence is sequentially updated by the coefficient correction amount calculating circuit 7 so as to minimize a residual signal obtained by subtracting pseudo acoustic echo, The acoustic impulse response characteristic output from the product-sum calculation circuit 13 is frequency-converted in the high-speed Fourier transform device 16 to directly calculate the spatial frequency transfer characteristic, and based on this value, the input / output evaluation device 17 zeros. Calculate the average distance between levels. Then, in the input / output compensation device 18,
A high-quality loudspeaker information communication system that determines the compensation level corresponding to the average value of the distance and inserts it into the communication line. Further, it is also possible to adopt a configuration in which the first high-quality loudspeaker information communication system and the second high-quality loudspeaker information communication system are combined. In this case, it is possible to configure high-quality voice communication in which no reverberation component is mixed and a loudspeaker information communication system in which the probability of howling is suppressed to an extremely low level. A high-quality loudspeaker information communication system configured to supply the impulse response characteristic and the spatial frequency transfer characteristic as video information to a user or an installer so that special settings can be made for each environment. The setting values can be determined using this evaluation signal even when the communication line is connected or disconnected, and can be determined automatically or manually. Of course, when the environment is not measured, the high-speed product-sum calculation circuit 13 is controlled so as to stop its operation and allow the transmission signal to pass through as it is.

【0013】図3は本発明で使用した評価信号の一例を
示した図である。この評価信号は時間引き延ばしパルス
(例えば、鈴木、浅野、曽根、”音響系の伝達関数の模
擬をめぐって(その2)”、日本音響学会誌 45, pp.46
-50.(1989))で評価信号としては代表的なものの一つで
ある。評価信号としては、精度が保証されていればどの
様なものでも良く、従って、ここで提示したものを使用
しなくとも本発明は構成できる。図4、図5、および図
6に示したのは各段階での演算結果である。図4は該高
速積和演算回路13の出力で、音響インパルス応答特性
である。この波形を該インパルス応答特性評価装置14
で自乗変化量を算出した結果を示したものが、図5であ
る。この評価値からは該可変係数ディジタルフィルタ3
の仕様を越える適応不可能な領域の残響量を求め、回路
損失量の指標とする。又、該音響インパルス応答特性の
波形を該高速フ−リエ変換装置16で周波数変換した空
間周波数伝達特性を示したのが図6である。図6の波形
から該入出力評価装置17は零レベルがらの距離の平均
値を算出し、該入出力補償装置18にその値を渡す。
FIG. 3 is a diagram showing an example of the evaluation signal used in the present invention. This evaluation signal is a time-delayed pulse (for example, Suzuki, Asano, Sone, "Simulation of transfer function of acoustic system (2)", The Acoustical Society of Japan, 45, pp.46.
-50. (1989)) is one of the typical evaluation signals. As the evaluation signal, any signal may be used as long as its accuracy is guaranteed. Therefore, the present invention can be configured without using the one presented here. FIG. 4, FIG. 5, and FIG. 6 show the calculation results at each stage. FIG. 4 shows an output of the high-speed product-sum calculation circuit 13, which is an acoustic impulse response characteristic. This waveform is used as the impulse response characteristic evaluation device 14
FIG. 5 shows the result of calculation of the squared variation amount at. From this evaluation value, the variable coefficient digital filter 3
The amount of reverberation in the non-adaptable area that exceeds the specifications of is calculated and used as an index of the amount of circuit loss. Further, FIG. 6 shows a spatial frequency transfer characteristic in which the waveform of the acoustic impulse response characteristic is frequency-converted by the high-speed Fourier transform device 16. From the waveform of FIG. 6, the input / output evaluation device 17 calculates the average value of the distances from the zero level and passes the value to the input / output compensation device 18.

【0014】[0014]

【発明の効果】以上、詳細に説明したように本発明によ
れば、下記のような効果が期待される。 (1)音響反響除去装置の適応性能を超える様な大きな
残響時間を有する環境下においても違和感のない高品質
な拡声音声情報通信システムを実現できる。
As described above in detail, according to the present invention, the following effects are expected. (1) It is possible to realize a high-quality loudspeaker information communication system that does not cause discomfort even in an environment having a large reverberation time that exceeds the adaptive performance of the acoustic echo canceller.

【0015】(2)結果的にの損失を通信回線に挿入し
なくても良くなるので、回線のレベルが小さくて済み、
音質の劣化を防ぐ事が可能である。
(2) Since the resultant loss does not have to be inserted into the communication line, the line level can be small,
It is possible to prevent deterioration of sound quality.

【0016】(3)通常、音響反響除去処理が使用され
る外部環境はその音響反響除去処理の性能を遥かに越え
ている場合が殆どである。その様な状況下でも本発明を
用いる事で、ハウリングの発生確率を極めて低く抑圧さ
せた高品質拡声情報通信システムを供給する事が出来
る。
(3) In most cases, the external environment in which the acoustic echo removal processing is used far exceeds the performance of the acoustic echo removal processing. Even in such a situation, by using the present invention, it is possible to provide a high-quality loudspeaker information communication system in which the probability of howling is suppressed to an extremely low level.

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

【図1】本発明による第一の高品質拡声情報通信システ
ムの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 a first high-quality loudspeaker information communication system according to the present invention.

【図2】本発明による第二の高品質拡声情報通信システ
ムの構成例を示すブロック図である。
FIG. 2 is a block diagram showing a configuration example of a second high-quality loudspeaker information communication system according to the present invention.

【図3】本発明の説明で用いた評価信号の時間引き延ば
しパルス波形の一例を示す図である。
FIG. 3 is a diagram showing an example of a time-extended pulse waveform of an evaluation signal used in the description of the present invention.

【図4】本発明の説明で用いた音響インパルス応答特性
の一例を示す図である。
FIG. 4 is a diagram showing an example of acoustic impulse response characteristics used in the description of the present invention.

【図5】本発明の説明で用いた音響インパルス応答波形
の自乗変化量の一例を示す図である。
FIG. 5 is a diagram showing an example of a square change amount of an acoustic impulse response waveform used in the description of the present invention.

【図6】本発明の説明で用いた音響インパルス応答波形
の周波数変換した空間周波数伝達特性の一例を示す図で
ある。
FIG. 6 is a diagram showing an example of frequency-converted spatial frequency transfer characteristics of the acoustic impulse response waveform used in the description of the present invention.

【図7】従来技術による音響反響除去装置の基本構成例
を示すブロック図である。
FIG. 7 is a block diagram showing an example of the basic configuration of a conventional acoustic echo canceller.

【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 音響反響除去装置 12 評価信号発生装置 13 高速積和演算回路 14 インパルス応答特性評価装置 15 回路損失制御装置 16 高速フ−リエ変換装置 17 入出力評価装置 18 入出力補償装置 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 Acoustic echo canceller 12 Evaluation signal generator 13 High-speed product-sum calculation circuit 14 Impulse response characteristic evaluation device 15 Circuit loss control device 16 High-speed Fourier transform device 17 Input / output evaluation device 18 Input / output compensation device

Claims (3)

【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
去装置を用いた拡声情報通信システムにおいて、音響特
性評価信号を生起する評価信号発生装置と、該評価信号
発生装置から発生された該音響評価信号が音響空間を介
して該送話信号入力端子より入力される応答信号に畳み
込み演算処理を加えて音響空間のインパルス応答特性を
算出する高速積和演算回路と、該インパルス応答特性に
適応した回線損失を挿入することにより残響成分を除去
することを特徴とする拡声情報通信システム。
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 a loudspeaker information communication system using an acoustic echo canceller in which a coefficient sequence is sequentially updated by a coefficient correction amount calculation circuit so as to minimize a residual signal to be obtained, an evaluation signal generation device that generates an acoustic characteristic evaluation signal, The acoustic evaluation signal generated from the evaluation signal generator is subjected to convolution calculation processing to the response signal input from the transmission signal input terminal via the acoustic space. A high-speed product-sum calculation circuit for calculating impulse response characteristics of an acoustic space, and a reverberation component is removed by inserting a line loss adapted to the impulse response characteristics.
【請求項2】該インパルス応答特性を周波数変換する高
速フ−リエ変換装置と、該高速フ−リエ変換装置からの
出力となる空間周波数伝達特性から平均空間損失を算出
する入出力評価装置とから構成され、該平均空間損失を
制御値として通信回線の入出力レベルを決定することに
よりハウリング発生の確率を低減させたことを特徴とす
る請求項1記載の拡声情報通信システム。
2. A high-speed Fourier transform device for frequency-converting the impulse response characteristic, and an input / output evaluation device for calculating an average spatial loss from the spatial-frequency transfer characteristic output from the high-speed Fourier transform device. The loudspeaker information communication system according to claim 1, wherein the probability of occurrence of howling is reduced by determining the input / output level of the communication line using the average space loss as a control value.
【請求項3】該インパルス応答特性と、該空間周波数伝
達特性とを映像情報として使用者または設置者に供給す
ることで該回線損失の大きさ等の制御値を調整できるこ
とを可能とし、より高品質な音声情報通信を実現した請
求項1記載の拡声情報通信システム。
3. A control value such as the magnitude of the line loss can be adjusted by supplying the impulse response characteristic and the spatial frequency transfer characteristic as image information to a user or an installer, and a higher value can be obtained. The loudspeaker information communication system according to claim 1, which realizes quality voice information communication.
JP24928495A 1995-09-27 1995-09-27 Loudspeaking information communication system Pending JPH0993166A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP24928495A JPH0993166A (en) 1995-09-27 1995-09-27 Loudspeaking information communication system

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP24928495A JPH0993166A (en) 1995-09-27 1995-09-27 Loudspeaking information communication system

Publications (1)

Publication Number Publication Date
JPH0993166A true JPH0993166A (en) 1997-04-04

Family

ID=17190684

Family Applications (1)

Application Number Title Priority Date Filing Date
JP24928495A Pending JPH0993166A (en) 1995-09-27 1995-09-27 Loudspeaking information communication system

Country Status (1)

Country Link
JP (1) JPH0993166A (en)

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