JP3714005B2 - Loudspeaker - Google Patents

Description

【００２１】
なお、上記式のＬ(s)は上記閉ループ系における一巡伝達関数、ｓはラプラス変数をそれぞれ表す。
【００２２】
ここで、閉ループ系の安定性は上記一巡伝達関数Ｌ(s)により判別することができる。すなわち、極座標系における一巡伝達関数Ｌ(s)のθ成分（＝∠Ｌ(s)）が∠Ｌ(s)＝２ｎπ（ｎは整数）となる全ての周波数において、ｒ成分（＝｜Ｌ(s)｜）が｜Ｌ(s)｜＜１ならば閉ループ系は安定、｜Ｌ(s)｜≧１となる周波数が存在すれば閉ループ系は不安定となり、その周波数において発振してハウリングが生じる。また、閉ループ系が安定である場合に、∠Ｌ(s)＝２ｎπとなる全ての周波数における利得の最大値をＬ_MAXとすれば、閉ループ系の利得余裕値は１／Ｌ_MAXで表される。よって、閉ループ系の安定性の尺度は閉ループ利得余裕値により評価することができる。
【００２３】
一方、閉ループ利得余裕値は、閉ループ系のインパルス応答特性と密接な関係があり、閉ループ利得余裕値が大きいほどインパルス応答信号Ｑの振幅が時間とともに急激に減衰し、閉ループ利得余裕値が小さいほど減衰が緩やかになる。そこで、閉ループ利得余裕推定部１２において閉ループ利得余裕値を推定するためのサンプル信号（インパルス信号）Ｐを上記閉ループ系に与えたときの応答信号Ｑを観測し、その応答信号Ｑの包絡線成分から閉ループ系の時定数に関する情報を抽出して閉ループ系におけるハウリング発生限界までの閉ループ利得余裕値の推定を行うとともに、推定結果に基づき、挿入損失量調整手段８にて上記所望の閉ループ利得余裕値に対する実際の挿入損失量の過不足分を算出して閉ループ系への挿入損失量を調整する。
【００２４】
すなわち、上述のように包絡線検波器１１で得られる応答信号Ｑの包絡線成分の時間特性が閉ループ利得余裕値が大きいほど減衰が早く且つ小さいほど減衰が緩やかになるという性質を有することから、閉ループ利得余裕推定部１２において事前に学習された種々の利得余裕値に対する包絡線検波器１１の出力データから閾値レベルを求めておき、比較器１３において観測される包絡線検波器１１の出力信号レベルを上記閾値レベルと比較することにより、その比較結果に基づいて判定部１４にて閉ループ利得余裕値が推定できる。そして、その推定結果から、閉ループ利得余裕値を設計仕様で定めた値とするために必要な損失量を挿入するべく、制御器７に信号を伝送して制御器７によって減衰器６₁，６₂の減衰量を調節している。
【００２５】
上述のように上記拡声通話機では、インパルス信号に対する応答信号から、閉ループ系での利得余裕値を推定し、この利得余裕値の推定値と、目標とする利得余裕値との差分を算出し、その算出結果に基づいて信号経路に挿入する損失量を調整する挿入損失量調整手段８を備えているので、通話中においても閉ループ系の安定度に応じて挿入損失量を制御し、常に利得余裕値を仕様で定めた値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができる。また、従来例に比較して必要以上に挿入損失量を大きくする必要がないため、双方向同時通話性能の実現可能性が高まるという利点もある。
【００２６】
なお、上記基本構成並びに後述する各実施形態においてはサンプル信号発生器にインパルス信号発生器９を用いたが、代わりにバースト信号発生器を用いてバースト信号をサンプル信号に用いてもよい。この場合には、信号を送出している状態から送出を停止した瞬間からの閉ループ系の過渡応答を観測し、その包絡線成分から閉ループ系の時定数に関する情報を抽出する。而してサンプル信号にバースト信号を用いた場合にも、信号送出時間が閉ループ系の遅延時間に対して十分に短ければ、その応答波形の包絡線成分より閉ループ系の時定数に関する情報を抽出することができ、バースト信号の信号送出時間を適切な値とすることにより通話中における違和感をなくし、サンプル信号の送出による通話品質の劣化を抑えることができるという利点がある。但し、種々の利得余裕値に対する閾値レべルを求めておく必要があることはインパルス信号の場合と同様である。
【００２７】
（実施形態１）
ところで、上記基本構成においては式１で表される応答信号Ｑから利得余裕値を推定しているが、上記式１からも明らかなように応答信号Ｑにはサンプル信号Ｐのみに対する応答信号（式１の右辺第１項）以外の信号が含まれている。そのため、受話信号Ｙ及びマイクロホン１の集音する音響信号Ｘのレベルが大きく、式１の右辺第２項及び第３項が無視できない場合においては、利得余裕値を精度良く推定することが難しくなる。
【００２８】
そこで、本発明の各実施形態においては、式１で表される応答信号Ｑのサンプル信号に対応する信号（式１の右辺第１項）のみを抽出することにより、その他の信号（式１の右辺第２項及び第３項）を充分に抑圧し、精度良く利得余裕値を推定することができるようにしている。
【００２９】
図１は本実施形態における挿入損失量調整手段８の構成を示すブロック図である。なお、本実施形態の拡声通話機の全体構成は上述の基本例と共通であるから図示並びに説明は省略し、共通する部分については同一の符号を付す。図１に示すように本実施形態における挿入損失量調整手段８は、インパルス信号発生器９、加算器１０、包絡線検波器１１並びに閉ループ利得余裕推定部１２に加えて、所定の時間間隔でインパルス信号発生器９から信号経路へ複数回にわたってインパルス信号を送出させるためのタイミング制御を行う同期制御部１５と、包絡線検波器１１の前段に設けられ同期制御部１５からのタイミング情報を受けてインパルス信号に対する応答信号Ｑを同期加算平均処理する同期加算平均部１６とを具備している。
【００３０】
図２は同期制御部１５並びに同期加算平均部１６を示すブロック図である。同期制御部１５は、図３に示すように所定の時間間隔Ｔ₁，Ｔ₂，…，Ｔ_N-1毎に複数回励振されるインパルス信号（サンプル信号）Ｐ₀，Ｐ₁，…，Ｐ_N-1（以下、｛Ｐ｝と表記する。）をインパルス信号発生器９から送出させるとともに、図２に示すようにインパルス信号Ｐ₀…の送出に同期した同期信号を同期加算平均部１６に出力する。
【００３１】
一方、同期加算平均部１６は、図２に示すように同期制御部１５からの同期信号によってオン・オフされるスイッチ要素１６ａと、スイッチ要素１６ａを介して入力されるインパルス信号｛Ｐ｝の応答信号｛Ｑ｝を所定の時間だけ遅延させる遅延器１６ｂ₁〜１６ｂ_N-1と、各遅延器１６ｂ₁〜１６ｂ_N-1の出力を加算する加算器１６ｃ₁〜１６ｃ_N-1と、全遅延器１６ｂ₁〜１６ｂ_N-1の出力信号の和（加算器１６ｃ_N-1の出力）を同期加算回数Ｎで除算する除算器１６ｄとを備えて、インパルス信号｛Ｐ｝に同期して加算平均処理を行って応答信号｛Ｑ｝の移動平均を求めている。而して、音響信号Ｘや受話信号Ｙの位相はインパルス信号｛Ｐ｝が送出される時間間隔Ｔ₁，Ｔ₂，…，Ｔ_N-1とは無関係であるため、同期加算回数Ｎを充分に大きくとることで式１における右辺第２項及び第３項が抑制されるとともに右辺第１項のインパルス信号｛Ｐ｝に対する応答成分のみが強調され、実質的に式１の右辺第１項の上記応答成分のみを抽出することが可能となる。なお、応答信号｛Ｑ｝の移動平均が求められたら、同期制御部１５から出力されるリセット信号により、同期加算平均部１６の各遅延器１６ｂ₁〜１６ｂ_N-1がリセットされ、同期加算平均部１６が新たに移動平均の算出可能な初期状態に復帰する。
【００３２】
そして、同期加算平均部１６から出力される応答信号｛Ｑ｝の移動平均が包絡線検波器１１に入力され、同期加算平均処理により得られたインパルス信号｛Ｐ｝に対する閉ループ系の応答信号の包絡線成分が抽出される。さらに、閉ループ利得余裕推定部１２では、包絡線検波器１１で抽出された包絡線成分の時間軸に対する減衰特性から利得余裕値を推定し、推定された利得余裕値に基づいて所要の挿入損失量を算出する。なお、インパルス信号｛Ｐ｝の時間間隔Ｔ₁，Ｔ₂，…，Ｔ_N-1は、利得余裕値の推定処理に必要な応答時間τに対して充分に大きくする必要がある。また、利得余裕値の推定処理に要する加算器１１の出力信号の観測時間Ｔ（＝Ｔ₀＋Ｔ₁＋…＋Ｔ_N-1＋τ）は、この時間Ｔ内における閉ループ系の特性の変動が無視できる程度に短くする必要がある。
【００３３】
上述のように本実施形態では、基本構成の挿入損失量調整手段８に同期制御部１５と同期加算平均部１６とを付加して、インパルス信号｛Ｐ｝に対する応答信号｛Ｑ｝の移動平均を求め、求めた移動平均を包絡線検波器１１に出力するようにしているため、閉ループ中に周囲騒音や音声信号が存在する場合においても、応答信号｛Ｑ｝に対する周囲騒音の影響を軽減し、利得余裕値を精度良く推定することができる。その結果、上記のような場合においても挿入損失量を適切な値に設定して、閉ループ系を安定化することができる。特に、本手法では、閉ループ中に存在する周囲騒音が広い周波数帯域を持ち、自己相関が少ない場合にその効果が大きい。
【００３４】
（実施形態２）
ところで、実施形態１においては閉ループ系の利得余裕値を推定するためにサンプル信号（インパルス信号又はバースト信号）を系に与えているが、通話中にインパルス信号やバースト信号がスピーカ３から聞こえるために使用者に聴感上の不快感を与えてしまう虞がある。
【００３５】
そこで本実施形態は、インパルス信号やバースト信号のように聴感上不快感を与える虞があるサンプル信号を使う代わりに、疑似白色信号を用いて利得余裕値を推定することによって、上述のような聴感上の不快感を与えるのを防止している。
【００３６】
図４は本実施形態の挿入損失量調整手段８を示すブロック図であり、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器２４と、音声信号に疑似白色信号を加算する加算器１０と、疑似白色信号と当該疑似白色信号に対する応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部２５と、相互相関演算部２５により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器１１と、包絡線検波器１１の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部１２とを具備している。なお、実施形態１と共通する部分については同一の符号を付して詳細な説明は省略する。
【００３７】
ここで、疑似白色信号としては、例えばＭ系列信号を用いる。Ｍ系列信号とは、次式のｆ(x)がｍｏｄ２の原始多項式であるとき、
ｆ(x)＝１＋ｃ₁ｘ＋ｃ₂ｘ²＋…ｃ_pｘ^p
次式により生成されるｍｏｄ２上の数列ａ_tで表される。
【００３８】
ａ_t＝ｃ₁ａ_t-1＋ｃ₂ａ_t-2＋…＋ｃ_pａ_t-p
また、この周期Ｔは２^p−１である。信号経路へ送出する信号ｐ(t)がＭ系列信号であり、その周期Ｔが充分に長ければ、Ｍ系列信号ｐ(t)に対して次式が成り立つ。
【００３９】
【式２】

【００４０】
ここで、Ψ_ppはＭ系列信号ｐ(t)の自己相関値を表すが、Ｍ系列信号の自己相関値が時間に依存しないため、次式のように表現することができる。
【００４１】
Ψ_pp(t、t＋τ)＝Ψ_pp(τ)
相互相関演算部２５においては、次式によりＭ系列信号ｐ(t)と加算器１０の出力信号（応答信号）ｑ(t)との相互相関値を上記Ａで規格化した値を求める。
【００４２】
【式３】

【００４３】
ここで、加算器１０の出力信号ｑ(t)は次式で表される。
【００４４】
【式４】

【００４５】
但し、ｈ_p(σ)、ｈ_y(σ)並びにｈ_x(σ)は各々Ｍ系列信号ｐ(t)の入力点から加算器１０の出力点への閉ループ系のインパルス応答、受話信号Ｙの入力点から加算器１０の出力点への閉ループ系のインパルス応答、送話信号Ｘの入力点から加算器１０の出力点への閉ループ系のインパルス応答をそれぞれ表している。上記式４を式３に代入すると次式が得られる。
【００４６】
【式５】

【００４７】
上記式５は次式のように変形できる。
【００４８】
【式６】

【００４９】
但し、Ψ_py及びΨ_pxはそれぞれＭ系列信号ｐ(t)と受話信号Ｙとの相互相関値、Ｍ系列信号ｐ(t)と送話信号Ｘとの相互相関値を表し、各々次式で与えられる。
【００５０】
【式７】

【００５１】
【式８】

【００５２】
ここで、Ｍ系列信号ｐ(t)と受話信号Ｙ、並びにＭ系列信号ｐ(t)と送話信号Ｘとは互いに因果関係がないため、それぞれの相互相関値Ψ_py、Ψ_pxは時間ｔに依存して変化する。また、Ｍ系列信号ｐ(t)の周期Ｔが充分に長いときには上記相互相関値Ψ_py、Ψ_pxはゼロとなる。よって、この場合には上記式６の右辺第２項及び第３項がゼロとなり、次式のように変形できる。
【００５３】
【式９】

【００５４】
従って、Ψ_pp(t、t＋τ)＝Ψ_pp(τ)で与えられるＭ系列信号（疑似白色信号）ｐ(t)と応答信号ｑ(t)との相互相関値を上記Ａで規格化した値を算出することにより、Ｍ系列信号ｐ(t)から応答信号ｑ(t)への系のインパルス応答の遅延時間τにおける係数を求めることができる。但し、上記式９が成り立つのは、式７及び式８で与えられる相互相関値Ψ_py，Ψ_pxがゼロとみなせるほどＭ系列信号ｐ(t)の周期Ｔが充分に長い場合であり、それ以外の場合においては、上記式６の右辺第２項及び第３項は外乱成分となり、インパルス応答の推定精度を劣化させる要因となる。
【００５５】
ところで、相互相関演算部２５をデジタル回路により実現する場合には、上記Ψ_pp(t、t＋τ)＝Ψ_pp(τ)を上記Ａで規格化した相互相関値は次式で表される
【００５６】
【式１０】

【００５７】
但し、Ｉは離散時間系におけるＭ系列信号ｐ(t)の周期を表し、ｐ(i)、ｑ(i+j)は各々サンプル時刻ｉ，ｉ＋ｊにおけるＭ系列信号ｐ(t)及び応答信号ｑ(t)のデータを表す。また、上記式７により得られる値はＭ系列信号ｐ(t)から応答信号ｑ(t)への系のインパルス応答の１係数ｈ_p(j)であり、閉ループ系の利得余裕を推定するために必要なインパルス応答観測時間をサンプル時間Ｊとすると、ｈ_p(0)，ｈ(1)，ｈ(2)，…，ｈ(J-1)の合計Ｊ個の係数を求める必要がある。よって、本実施形態においては、Ｍ系列信号（疑似白色信号）ｐ(t)から応答信号ｑ(t)への系のインパルス応答を推定するために、Ｉ×Ｊ回の積和演算を必要とする。
【００５８】
上述のような演算処理により求められる疑似白色信号ｐ(t)から応答信号ｑ(t)への系のインパルス応答（応答信号Ｑ）が包絡線検波器１１に入力され、実施形態１で説明したように閉ループ利得余裕推定部１２において包絡線検波器１１で抽出された包絡線成分の時間軸に対する減衰特性から利得余裕値を推定し、推定された利得余裕値に基づいて所要の挿入損失量を算出する。
【００５９】
上述のように本実施形態では、基本構成の挿入損失量調整手段８に対してインパルス信号発生器９の代わりに疑似白色信号発生器２４を付加するとともに、疑似白色信号と当該疑似白色信号に対する応答信号の相互相関値を求めて相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部２５を付加して、推定したインパルス応答（応答信号）から包絡線検波器１１並びに閉ループ利得余裕推定部１２により利得余裕値を推定するため、使用者に聴感上の不快感を与えることなく利得余裕値を精度よく推定でき、挿入損失量を適切な値に設定して、閉ループ系を安定化することができる。
【００６０】
（実施形態３）
図５は本発明の実施形態３における挿入損失量調整手段８を示すブロック図であり、実施形態２に対して実施形態１で説明した同期制御部１５並びに同期加算平均部１６を設けた点に特徴がある。
【００６１】
本実施形態は、信号経路に送出する疑似白色信号の周期が充分に長くなく、上記式６の右辺第２項及び第３項がゼロとはみなせずに外乱要素となる場合に、同期加算平均処理によりこれらの影響を低減し、閉ループ系のインパルス応答の推定精度を向上させるものである。いま、Ｍ系列信号ｐ(t)の周期をＴ₁、同期加算回数をＮとすると、同期加算平均部１６の出力値は次式で表される。
【００６２】
【式１１】

【００６３】
上記式１１に式６及び式９を代入して整理すると次式が得られる。
【００６４】
【式１２】

【００６５】
ここで、相互相関値Ψ_py，Ψ_pxは時間ｔに関して不規則に変動するため、上記式１２における右辺第２項及び第３項が同期加算回数Ｎの値を充分に大きくすることにより抑圧することができる。
【００６６】
本実施形態によれば、信号経路に送出するＭ系列信号ｐ(t)の周期Ｔを実施形態２に比較して短くすることができるため、挿入損失量調整手段８をデジタル回路で実現する場合にＭ系列信号ｐ(t)の発生及び上記式１０の演算に必要となる記憶装置の容量や積和回数を低減することができるという利点がある。なお、同期加算平均部１６より出力される閉ループ系のインパルス応答（応答信号）は包絡線検波器１１に入力され、実施形態１と同様の演算処理によって所要の挿入損失量が算出される。
【００６７】
（実施形態４）
図６は本発明の実施形態４における挿入損失量調整手段８を示すブロック図であり、実施形態１に対して騒音レベル推定部２０とインパルス信号送出レベル調整部２１とを設けた点に特徴がある。
【００６８】
騒音レベル推定部２０は、インパルス信号｛Ｐ｝の閉ループ系への送出以前に閉ループ中に存在する周囲騒音のレベルを推定するものであり、インパルス信号送出レベル調整部２１は、推定された周囲騒音のレベルに応じてインパルス信号発生器９から出力されるインパルス信号Ｐ₀…のレベルを調整するものである。すなわち、本発明に係る利得余裕値の推定処理においては、閉ループ中の周囲騒音レベルに対するインパルス信号送出レベルの比（この比をここではＳ／Ｎ比と定義する）が利得余裕値の推定精度に影響を及ぼすことから、閉ループにインパルス信号Ｐ₀…が送出される以前の周囲騒音レベルを騒音レベル推定部２０で推定し、推定された周囲騒音レベルに対して上記Ｓ／Ｎ比が所定値以上となるようにインパルス信号送出レベル調整部２１にてインパルス信号Ｐ₀…の送出レベルを調整することで、利得余裕値の推定精度を向上させることができる。
【００６９】
上述のように本実施形態では、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部２０と、騒音レベル推定部２０による推定結果に基づいてサンプル信号（インパルス信号）の送出レベルを調整するインパルス信号送出レベル調整部２１とを挿入損失量調整手段８に設けたので、閉ループ系に送出するインパルス信号のレベルを周囲騒音レベルに対して充分な大きさに設定することにより、観測点において高いＳ／Ｎ比でインパルス信号｛Ｐ｝に対する応答信号｛Ｑ｝を得ることができ、利得余裕値の推定精度を向上させることができる。なお、実施形態３の構成に対して騒音レベル推定部２０並びに疑似白色信号の送出レベルを調整する手段を設けても同様の作用効果を奏することができる。
【００７０】
【発明の効果】
請求項１の発明は、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第１の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第２の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号に対する応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する応答信号を同期加算平均処理する同期加算平均部とを具備して成るので、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で２−４線変換を行う必要がある場合に用いられる２−４線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、サンプル信号を入力したときに観測される応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まるという効果がある。しかも、同期制御部においてサンプル信号の送出タイミングを制御し、同期加算平均部でサンプル信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に周囲騒音や音声信号が存在する場合であっても、これらの影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができるという効果がある。特に、本発明は、閉ループ中に存在する周囲騒音が広い周波数帯域を持ち、自己相関が少ない場合に効果が大きい。
【００７１】
請求項２の発明は、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第１の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第２の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号に対する応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部とを具備して成るので、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で２−４線変換を行う必要がある場合に用いられる２−４線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、周期が有限の疑似白色信号を入力したときに観測される応答信号と当該疑似白色信号との相互相関値を求めるとともに求めた相互相関値から推定される閉ループ系のインパルス応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まるという効果がある。しかも、利得余裕の推定に疑似白色信号を用いるので、使用者に聴感上の不快感を与えるのを防止することができるという効果がある。
【００７２】
請求項３の発明は、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部とを挿入損失量調整手段が具備して成るので、同期制御部において疑似白色信号の送出タイミングを制御し、同期加算平均部で疑似白色信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができるという効果がある。特に、本発明は、閉ループ中に存在する周囲騒音が広い周波数帯域を持ち、自己相関が少ない場合に効果が大きい。
【００７３】
請求項４の発明は、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいてサンプル信号又は疑似白色信号の送出レベルを調整する送出レベル調整部とを挿入損失量調整手段が具備して成るので、閉ループ系に送出するサンプル信号又は疑似白色信号のレベルを周囲騒音レベルに対して充分な大きさに設定することにより、観測点において高いＳ／Ｎ比でサンプル信号又は疑似白色信号に対する応答信号を得ることができ、利得余裕値の推定精度を向上させることができるという効果がある。
【図面の簡単な説明】
【図１】 実施形態１における挿入損失量調整手段を示すブロック図である。
【図２】 同上における同期制御部並びに同期加算平均部を示すブロック図である。
【図３】 同上の動作を説明するための説明図である。
【図４】 実施形態２における挿入損失量調整手段を示すブロック図である。
【図５】 実施形態３における挿入損失量調整手段を示すブロック図である。
【図６】 実施形態４における挿入損失量調整手段を示すブロック図である。
【図７】 （ａ）は本発明の基本構成を示すブロック図、（ｂ）は同じく挿入損失量調整手段の基本構成を示すブロック図である。
【図８】 同上における閉ループを説明するための説明図である。
【符号の説明】
８ 挿入損失量調整手段
９ インパルス信号発生器
１０ 加算器
１１ 包絡線検波器
１２ 閉ループ利得余裕推定部
１５ 同期制御部
１６ 同期加算平均部[0001]
BACKGROUND OF THE INVENTION
  The present invention relates to a loudspeaker used in homes, buildings, factories, and the like.
[0002]
[Prior art]
  Conventionally, in a loudspeaker such as an interphone, a telephone, or a PHS, a closed loop is formed by acoustic feedback from a speaker to a microphone and impedance mismatch in a hybrid circuit (2-4 wire conversion circuit), and the gain of the amplifier is large. Since howling occurs when the closed-loop gain becomes 1 or more for reasons such as too much, suppression of howling has become an indispensable issue in securing call quality.
[0003]
  Therefore, conventionally, a howling suppression method has been used in which a closed loop gain is suppressed by inserting a predetermined amount of loss into a received signal or a transmitted signal in accordance with the level of the transmitted / received signal. In addition, there is another method that uses an echo canceller, but it tends to be unstable when the echo path changes suddenly due to a transient state where the coefficient of the adaptive filter in the echo canceller has not converged or system fluctuations. Often used in conjunction with insertion loss.
[0004]
  In any of the above cases, if the amount of insertion loss is increased too much, the quality of the call will be degraded during a call (sound will be interrupted). Is desirable.
[0005]
  In response to such problems, the present inventors have already monitored the gain margin of the closed loop formed in the voice communication system as needed, and adaptively adjust the insertion loss amount based on the estimated gain margin value. (See Japanese Patent Application No. 9-61434).
[0006]
  FIG.(A) is a block diagram showing a loudspeaker according to the above application, and a microphone 1 that outputs the collected sound as a voice signal on the transmission side (hereinafter referred to as a transmission signal), and a transmission from the microphone 1. A first amplifier 2 that amplifies the speech signal, a speaker 3 that rings in response to a voice signal on the receiving side (hereinafter referred to as a received signal), and a second amplifier 4 that amplifies the received signal output to the speaker 3 And a hybrid circuit 5 as 2-4 line conversion means for performing 2-4 line conversion between the transmission side and reception side and an external communication line, and a predetermined amount of loss in the signal paths on the transmission side and reception side Attenuator 6 as loss insertion means for inserting₁, 6₂And each attenuator 6₁, 6₂A gain margin in a closed loop formed by acoustic coupling from the microphone 1 to the speaker 3 and reflection in the hybrid circuit 5 in accordance with a controller 7 that variably controls the loss amount of the signal and a response signal to the sample signal sent to the signal path. Insertion loss amount adjusting means 8 is provided for estimating and adjusting the loss amount based on the estimation result via the controller 7. The insertion loss amount adjusting means 8 can be provided on the signal path on the transmission side.
[0007]
  On the other hand, FIG. 4B is a block diagram showing a specific configuration of the insertion loss amount adjusting means 8, and an impulse signal generator 9 for sending an impulse signal as a sample signal to the signal path on the reception side, and the reception signal An adder 10 for adding the impulse signal to the envelope, an envelope detector 11 for detecting the envelope of the response signal to the impulse signal, and a closed loop gain for estimating the gain margin in the closed loop system based on the output signal of the envelope detector 11 And a margin estimation unit 12. Here, the closed loop gain margin estimation unit 12 compares the output signal level of the envelope detector 11 with a predetermined threshold level, and a determination unit that estimates the gain margin value of the closed loop system based on the comparison result. 14. The sample signal may be a single pulse signal with a sufficiently short pulse width or a burst signal (burst noise). AndFIG.As shown, the second amplifier 4 → speaker 3 → microphone 1 → first amplifier 2 → attenuator 6₁→ Hybrid circuit 5 → Attenuator 6₂→ Insertion loss adjustment means 8 → The second amplifier 4 forms a closed loop.
[0008]
  Thus, the output signal of the adder 10 is a response signal when a sample signal is input to the closed loop system, and the impulse response of the closed loop system can be estimated by observing the output signal of the adder 10. . The extracted envelope component is closely related to the real part of the dominant pole (the pole closest to the imaginary axis on the complex plane) in the transfer function of the closed-loop system. Stability, that is, gain margin can be estimated. However,FIG.In the output signal of the adder 10 in (a), the transmission / reception voice signal and the ambient noise are superimposed in addition to the response component for the sample signal, and only the response component for the sample signal is extracted to the insertion loss amount adjusting means 8. Therefore, there is a problem that the accuracy of estimating the gain margin value is deteriorated when the level of the voiced section and the ambient noise signal is large. Among these, for the deterioration of the estimation accuracy due to the transmitted / received speech, a silence detector for detecting a silent section during a call is provided from the level of the received signal (or may be the level of the transmitted signal). Only when a silent section is detected by the detector, a sample signal (impulse signal or burst noise) is input to the closed-loop system, and the closed-loop gain margin estimation unit 12 performs an estimation process of the gain margin value to increase the estimation accuracy. Can be improved.
[0009]
[Problems to be solved by the invention]
  However, when a steady ambient noise signal exists in the closed loop of the above-mentioned loudspeaker communication system and its level is large, the loudspeaker according to the above application cannot accurately estimate the gain margin value. . In addition, in the above method, since it is necessary to stop the gain margin value estimation process when a voice signal is present in the closed loop, this is handled when the gain margin value of the system fluctuates during a call. It may not be possible and the system may become unstable.
[0010]
  The present invention has been made in view of the above problems, and its object is to obtain a gain with sufficient accuracy even when ambient noise and audio signals are present in a closed loop and the levels of those signals are large. An object of the present invention is to provide a loudspeaker capable of estimating a margin value and always realizing a stable call.
[0011]
[Means for Solving the Problems]
  In order to achieve the above object, the first aspect of the present invention provides a microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And a control means for variably controlling the amount of loss inserted from the loss insertion means, and a control means for estimating a gain margin in a closed loop formed through a microphone and a speaker according to a response signal to the sample signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the insertion loss, and the insertion loss amount adjusting means sends a sample signal to the signal path. A sample signal generator, an adder that adds the sample signal to the audio signal, an envelope detector that detects the envelope of the response signal to the sample signal, and a gain margin in the closed loop based on the output signal of the envelope detector A closed loop gain margin estimation unit for estimation, a synchronization control unit for performing timing control for sending a sample signal multiple times from a sample signal generator to a signal path at a predetermined time interval, and a front stage of an envelope detector A synchronization addition averaging unit that receives the timing information from the synchronization control unit and performs a synchronous addition averaging process on a response signal to the sample signal, and includes an acoustic coupling from a speaker to a microphone and a transmission side and a reception side; Impedance of 2-4 wire conversion means used when 2-4 wire conversion is required with an external telephone line For a closed loop system that is formed due to reflection due to matching, the gain margin value is estimated based on the attenuation characteristics of the envelope component of the response signal observed when the sample signal is input, and the desired gain margin value is excessive or insufficient. Since the amount of insertion loss is adjusted by calculating the minutes, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality can be maintained without causing howling. In addition, since the amount of insertion loss is not increased more than necessary, the feasibility of bidirectional simultaneous call performance increases. Moreover, the synchronization control unit controls the transmission timing of the sample signal, the synchronous addition averaging unit performs the moving average process synchronized with the sample signal, and the envelope detector detects the envelope component from the output signal of the synchronous addition average unit. Because it is extracted, even when the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced, the gain margin value is estimated with sufficient accuracy, and the insertion loss amount can be set to an appropriate value Thus, a stable call can be realized at all times.
[0012]
  In order to achieve the above object, the invention according to claim 2 is a microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And a control means for variably controlling the amount of loss inserted from the loss insertion means, and a control means for estimating a gain margin in a closed loop formed through a microphone and a speaker according to a response signal to the sample signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the insertion loss, the insertion loss amount adjusting means is a pseudo white with a finite period in the signal path A pseudo white signal generator for inserting a signal, an adder for adding the pseudo white signal to the audio signal, a cross correlation value of the pseudo white signal and a response signal for the pseudo white signal and a cross correlation value obtained A cross-correlation operation unit that estimates the impulse response of the closed loop, an envelope detector that detects an envelope of the impulse response signal of the closed loop estimated by the cross-correlation operation unit, and a closed loop based on the output signal of the envelope detector A closed loop gain margin estimating unit for estimating a gain margin, and performing acoustic coupling from a speaker to a microphone, and performing 2-4 wire conversion between a transmitting side, a receiving side, and an external communication line The period is finite compared to the closed loop system formed due to reflection due to impedance mismatch in the 2-4 wire conversion means used when necessary Gain is obtained by the attenuation characteristic of the envelope component of the impulse response signal of the closed-loop system estimated from the cross-correlation value between the response signal observed when the pseudo-white signal is input and the pseudo-white signal. In order to adjust the insertion loss amount by estimating the margin value and calculating the excess and deficiency relative to the desired gain margin value, the gain margin of the closed loop system until howling can be maintained at the desired value, Stable call quality can be maintained without generating howling, and the amount of insertion loss is not increased more than necessary, thereby increasing the possibility of realizing bidirectional simultaneous call performance. In addition, since the pseudo white signal is used for the estimation of the gain margin, it is possible to prevent the user from feeling uncomfortable in hearing.
[0013]
  According to a third aspect of the present invention, in the second aspect of the present invention, there is provided a synchronization control unit for performing timing control for sending a pseudo white signal to a signal path from a pseudo white signal generator to a signal path at a predetermined time interval, and an envelope. The insertion loss amount adjusting means includes a synchronous addition averaging unit that is provided in a preceding stage of the detector and receives the timing information from the synchronization control unit and performs a synchronous addition averaging process on the closed loop impulse response signal estimated by the cross correlation calculation unit. The synchronous control unit controls the transmission timing of the pseudo white signal, the synchronous addition averaging unit performs the moving average process synchronized with the pseudo white signal, and the envelope detection from the output signal of the synchronous addition average unit Since the envelope component is extracted by the detector, the effect of ambient noise is reduced even when the level of stationary ambient noise existing in the closed loop is large. Estimating the gain margin value with sufficient accuracy, the insertion loss becomes possible to set to appropriate values, it is possible to always realize a stable call.
[0014]
  Claim4According to the invention of claim 1 or 3, in the invention of claim 1 or 3, a noise level estimator for estimating the level of ambient noise existing in the closed loop, and a transmission level of the sample signal or pseudo white signal based on the estimation result by the noise level estimator And a transmission level adjustment unit for adjusting the level of the sample signal or the pseudo white signal sent to the closed loop system is set to a sufficient level with respect to the ambient noise level. By doing so, it is possible to obtain a response signal to the sample signal or the pseudo white signal at a high S / N ratio at the observation point, and to improve the estimation accuracy of the gain margin value..
[0015]
DETAILED DESCRIPTION OF THE INVENTION
  Before describing the embodiment of the present invention, the configuration and operation of a loudspeaker according to a basic prior application (Japanese Patent Application No. 9-61434) will be described in detail.
[0016]
  As described in the prior art, the loudspeaker described in the above publication includes a microphone 1 that outputs a collected sound as a voice signal on the transmission side (hereinafter referred to as a transmission signal), and a microphone 1 A first amplifier 2 for amplifying a transmission signal, a speaker 3 that rings in response to a voice signal on the reception side (hereinafter referred to as a reception signal), and a second amplifier that amplifies a reception signal output to the speaker 3 4 and a hybrid circuit 5 as 2-4 line conversion means for performing 2-4 line conversion between the transmission side and reception side and an external communication line, and a predetermined amount of signal paths on the transmission side and reception side. Attenuator 6 as loss insertion means for inserting loss₁, 6₂And each attenuator 6₁, 6₂A controller 7 that variably controls the loss amount of the signal and a gain margin in a closed loop system formed through the microphone 1 and the speaker 3 in accordance with a response signal to the sample signal sent to the signal path and through the controller 7 And an insertion loss amount adjusting means 8 for adjusting the loss amount based on the estimation result.
[0017]
  On the other hand, the insertion loss amount adjusting means 8 includes an impulse signal generator 9 for sending an impulse signal as a sample signal to the signal path on the reception side, an adder 10 for adding the impulse signal to the reception signal, and a response signal for the impulse signal. An envelope detector 11 for detecting an envelope and a closed loop gain margin estimating unit 12 for estimating a gain margin in a closed loop system based on an output signal of the envelope detector 11 are configured by a microcomputer, a DSP (Digital Signal Processor), or the like. Is done. Here, the closed loop gain margin estimation unit 12 compares the output signal level of the envelope detector 11 with a previously obtained threshold level as described later, and calculates the gain margin value of the closed loop system based on the comparison result. And a determination unit 14 for estimation. The impulse signal used as the sample signal may be a single pulse signal with a sufficiently short pulse width. The envelope detector 11 can be composed of a synthesis circuit of a rectifier circuit and a low-pass filter circuit, a digital circuit such as a cyclic low-pass filter and a leak integrator, or a signal processing means such as a DSP (Digital Signal Processor).
[0018]
  Also,FIG.As shown, the second amplifier 4 → speaker 3 → microphone 1 → first amplifier 2 → attenuator 6₁→ Hybrid circuit 5 → Attenuator 6₂→ Insertion loss adjustment means 8 → The second amplifier 4 forms a closed loop. Here, the transfer function in each part is defined as follows.
[0019]
    S: Electromechanical conversion characteristics of speaker 3
    G: Sound transmission characteristics from the speaker 3 to the microphone 1
    M: Acoustoelectric conversion characteristics of microphone 1
    Kr: amplification characteristic of the second amplifier 4
    Kx: amplification characteristic of the first amplifier 2
    Ar: Attenuator 6 on the receiver side₂Damping characteristics
    Ax: Transmitter side attenuator 6₁Damping characteristics
    Γ: reflection transfer function in the hybrid circuit 5
  Further, the impulse signal output from the impulse signal generator 9 is P, the far-end speaker voice input signal transmitted from the external communication line is Y (hereinafter referred to as “received signal”), and the microphone 1 collects sound. When the sum of the near-end speaker voice signal and the ambient noise is X (hereinafter referred to as “acoustic signal”), the output signal (response) of the adder 10 which is one of the components of the insertion loss amount adjusting means 8 Signal) Q is expressed by the following equation.
[0020]
[Formula 1]

[0021]
  In the above equation, L (s) represents a one-round transfer function in the closed loop system, and s represents a Laplace variable.
[0022]
  Here, the stability of the closed-loop system can be discriminated by the one-round transfer function L (s). That is, the r component (= | L () at all frequencies where the θ component (= ∠L (s)) of the circular transfer function L (s) in the polar coordinate system is ∠L (s) = 2nπ (n is an integer). If s) |) is | L (s) | <1, the closed loop system is stable, and if there is a frequency where | L (s) | ≧ 1, the closed loop system becomes unstable and oscillates at that frequency and howling occurs. Arise. Further, when the closed loop system is stable, the maximum value of the gain at all frequencies where ∠L (s) = 2nπ is expressed as L_MAXThen, the gain margin value of the closed loop system is 1 / L_MAXIt is represented by Therefore, a measure of the stability of the closed loop system can be evaluated by the closed loop gain margin value.
[0023]
  On the other hand, the closed-loop gain margin value is closely related to the impulse response characteristics of the closed-loop system. The larger the closed-loop gain margin value, the more the amplitude of the impulse response signal Q attenuates with time, and the smaller the closed-loop gain margin value, the more attenuated. Becomes moderate. Accordingly, the closed loop gain margin estimation unit 12 observes the response signal Q when the sample signal (impulse signal) P for estimating the closed loop gain margin value is applied to the closed loop system, and the envelope signal component of the response signal Q is observed. Information on the time constant of the closed loop system is extracted to estimate the closed loop gain margin value up to the howling occurrence limit in the closed loop system, and based on the estimation result, the insertion loss amount adjusting means 8 is used to estimate the desired closed loop gain margin value. Calculate the amount of insertion loss to the closed loop system by calculating the excess and deficiency of the actual amount of insertion loss.
[0024]
  That is, since the time characteristic of the envelope component of the response signal Q obtained by the envelope detector 11 as described above has the property that the larger the closed loop gain margin value, the faster the attenuation and the smaller the attenuation, the slower the attenuation. A threshold level is obtained from the output data of the envelope detector 11 for various gain margin values learned in advance by the closed loop gain margin estimation unit 12, and the output signal level of the envelope detector 11 observed by the comparator 13. Is compared with the threshold level, the determination unit 14 can estimate the closed-loop gain margin based on the comparison result. Then, from the estimation result, a signal is transmitted to the controller 7 and the attenuator 6 is transmitted by the controller 7 in order to insert a loss amount necessary to make the closed loop gain margin value a value determined by the design specification.₁, 6₂The amount of attenuation is adjusted.
[0025]
  As described above, in the loudspeaker, the gain margin value in the closed loop system is estimated from the response signal to the impulse signal, the difference between the estimated gain margin value and the target gain margin value is calculated, Since the insertion loss amount adjusting means 8 for adjusting the loss amount inserted into the signal path based on the calculation result is provided, the insertion loss amount is controlled according to the stability of the closed loop system even during a call, and the gain margin is always obtained. The value can be maintained at a value determined by the specification, and stable call quality can be maintained without causing howling. Moreover, since it is not necessary to increase the amount of insertion loss more than necessary as compared with the conventional example, there is an advantage that the possibility of realizing bidirectional simultaneous call performance is increased.
[0026]
  Although the impulse signal generator 9 is used as the sample signal generator in the basic configuration and each of the embodiments described later, a burst signal may be used as the sample signal by using a burst signal generator instead. In this case, the transient response of the closed loop system from the moment when the transmission is stopped from the state in which the signal is transmitted is observed, and information on the time constant of the closed loop system is extracted from the envelope component. Thus, even when a burst signal is used as the sample signal, if the signal transmission time is sufficiently shorter than the delay time of the closed loop system, information on the time constant of the closed loop system is extracted from the envelope component of the response waveform. In addition, there is an advantage that by making the signal transmission time of the burst signal an appropriate value, it is possible to eliminate a sense of incongruity during a call and to suppress deterioration in call quality due to the transmission of a sample signal. However, as in the case of the impulse signal, it is necessary to obtain threshold levels for various gain margin values.
[0027]
  (Embodiment 1)
  In the above basic configuration, the gain margin value is estimated from the response signal Q expressed by Equation 1, but as is clear from Equation 1, the response signal Q includes a response signal (formula 1 except for the first term on the right side of 1). Therefore, when the level of the received signal Y and the sound signal X collected by the microphone 1 is large and the second term and the third term on the right side of Equation 1 cannot be ignored, it is difficult to accurately estimate the gain margin value. .
[0028]
  Therefore, in each embodiment of the present invention, by extracting only the signal corresponding to the sample signal of the response signal Q represented by Expression 1 (the first term on the right side of Expression 1), other signals (Expression 1 The second term and the third term on the right side are sufficiently suppressed so that the gain margin value can be estimated with high accuracy.
[0029]
  FIG. 1 is a block diagram showing a configuration of the insertion loss amount adjusting means 8 in the present embodiment. In addition, since the whole structure of the loudspeaker of this embodiment is common with the above-mentioned basic example, illustration and description are abbreviate | omitted and the same code | symbol is attached | subjected about a common part. As shown in FIG. 1, in addition to the impulse signal generator 9, the adder 10, the envelope detector 11, and the closed loop gain margin estimation unit 12, the insertion loss amount adjusting means 8 in the present embodiment includes impulses at predetermined time intervals. A synchronization control unit 15 that performs timing control for sending an impulse signal from the signal generator 9 to the signal path a plurality of times, and an impulse that is provided in the preceding stage of the envelope detector 11 and receives timing information from the synchronization control unit 15 And a synchronous addition averaging unit 16 for performing a synchronous addition averaging process on the response signal Q to the signal.
[0030]
  FIG. 2 is a block diagram showing the synchronization control unit 15 and the synchronization addition averaging unit 16. As shown in FIG. 3, the synchronization control unit 15 performs a predetermined time interval T.₁, T₂, ..., T_N-1Impulse signal (sample signal) P excited multiple times for each₀, P₁, ..., P_N-1(Hereinafter referred to as {P}) is sent out from the impulse signal generator 9 and the impulse signal P as shown in FIG.₀The synchronization signal synchronized with the transmission of... Is output to the synchronization addition averaging unit 16.
[0031]
  On the other hand, as shown in FIG. 2, the synchronous addition averaging unit 16 is turned on / off by the synchronization signal from the synchronization control unit 15, and the response of the impulse signal {P} input through the switch element 16a. Delay device 16b for delaying signal {Q} by a predetermined time₁~ 16b_N-1And each delay unit 16b₁~ 16b_N-116c for adding the outputs of₁~ 16c_N-1And all delay devices 16b₁~ 16b_N-1Output signal sum (adder 16c_N-1And a divider 16d that divides the output by the number N of synchronous additions, and performs an averaging process in synchronization with the impulse signal {P} to obtain a moving average of the response signal {Q}. Thus, the phase of the acoustic signal X and the reception signal Y is the time interval T at which the impulse signal {P} is transmitted.₁, T₂, ..., T_N-1Since the number N of synchronous additions is sufficiently large, the second and third terms on the right side in Equation 1 are suppressed and only the response component for the impulse signal {P} in the first term on the right side is emphasized. Thus, substantially only the response component of the first term on the right side of Equation 1 can be extracted. When the moving average of the response signal {Q} is obtained, each delay device 16b of the synchronous addition averaging unit 16 is reset by a reset signal output from the synchronization control unit 15.₁~ 16b_N-1Is reset, and the synchronous addition averaging unit 16 returns to an initial state in which a moving average can be newly calculated.
[0032]
  Then, the moving average of the response signal {Q} output from the synchronous addition averaging unit 16 is input to the envelope detector 11, and the envelope of the closed loop response signal with respect to the impulse signal {P} obtained by the synchronous addition averaging process Line components are extracted. Further, the closed loop gain margin estimation unit 12 estimates a gain margin value from the attenuation characteristic of the envelope component extracted by the envelope detector 11 with respect to the time axis, and a required insertion loss amount based on the estimated gain margin value. Is calculated. The time interval T of the impulse signal {P}₁, T₂, ..., T_N-1Needs to be sufficiently larger than the response time τ required for the process of estimating the gain margin value. Further, the observation time T (= T of the output signal of the adder 11 required for the process of estimating the gain margin value₀+ T₁+ ... + T_N-1+ Τ) needs to be shortened to such an extent that fluctuations in the characteristics of the closed loop system within this time T can be ignored.
[0033]
  As described above, in this embodiment, the synchronization control unit 15 and the synchronization addition averaging unit 16 are added to the insertion loss amount adjusting unit 8 having the basic configuration, and the moving average of the response signal {Q} with respect to the impulse signal {P} is obtained. Since the obtained moving average is output to the envelope detector 11, the influence of the ambient noise on the response signal {Q} is reduced even when ambient noise or a voice signal is present in the closed loop. The gain margin value can be estimated with high accuracy. As a result, even in the above case, the closed-loop system can be stabilized by setting the insertion loss amount to an appropriate value. In particular, this method is particularly effective when the ambient noise existing in the closed loop has a wide frequency band and the autocorrelation is small.
[0034]
  (Embodiment 2)
  By the way, in the first embodiment, a sample signal (impulse signal or burst signal) is given to the system in order to estimate the gain margin value of the closed loop system, but the impulse signal or burst signal can be heard from the speaker 3 during a call. There is a risk that the user may feel uncomfortable hearing.
[0035]
  Therefore, in this embodiment, instead of using a sample signal that may cause hearing discomfort such as an impulse signal or a burst signal, a gain margin value is estimated using a pseudo-white signal, and thus the above-described audibility is obtained. Preventing the above discomfort.
[0036]
  FIG. 4 is a block diagram showing the insertion loss amount adjusting means 8 of this embodiment. The pseudo white signal generator 24 inserts a pseudo white signal having a finite period into the signal path, and adds the pseudo white signal to the audio signal. An adder 10, a cross-correlation calculation unit 25 that calculates a cross-correlation value of the pseudo-white signal and a response signal to the pseudo-white signal, and estimates a closed-loop impulse response based on the calculated cross-correlation value; An envelope detector 11 for detecting the envelope of the closed-loop impulse response signal estimated by the above, and a closed-loop gain margin estimator 12 for estimating the gain margin in the closed-loop based on the output signal of the envelope detector 11. ing. In addition, about the part which is common in Embodiment 1, the same code | symbol is attached | subjected and detailed description is abbreviate | omitted.
[0037]
  Here, for example, an M-sequence signal is used as the pseudo white signal. The M-sequence signal means that f (x) in the following equation is a primitive polynomial of mod 2
      f (x) = 1 + c₁x + c₂x²+ ... c_px^p
A sequence a on mod2 generated by the following equation_tIt is represented by
[0038]
      a_t= C₁a_t-1+ C₂a_t-2+ ... + c_pa_tp
The period T is 2^p-1. If the signal p (t) sent to the signal path is an M-sequence signal and its period T is sufficiently long, the following equation holds for the M-sequence signal p (t).
[0039]
[Formula 2]

[0040]
Where Ψ_ppRepresents the autocorrelation value of the M-sequence signal p (t). Since the autocorrelation value of the M-sequence signal does not depend on time, it can be expressed as the following equation.
[0041]
      Ψ_pp(t, t + τ) = Ψ_pp(τ)
  In the cross-correlation calculating unit 25, a value obtained by normalizing the cross-correlation value between the M-sequence signal p (t) and the output signal (response signal) q (t) of the adder 10 with the above-described A is obtained.
[0042]
[Formula 3]

[0043]
Here, the output signal q (t) of the adder 10 is expressed by the following equation.
[0044]
[Formula 4]

[0045]
However, h_p(σ), h_y(σ) and h_x(σ) is a closed-loop impulse response from the input point of the M-sequence signal p (t) to the output point of the adder 10, and a closed-loop impulse response from the input point of the reception signal Y to the output point of the adder 10, respectively. , The impulse response of the closed loop system from the input point of the transmission signal X to the output point of the adder 10 is shown. Substituting Equation 4 into Equation 3 yields:
[0046]
[Formula 5]

[0047]
The above equation 5 can be transformed as the following equation.
[0048]
[Formula 6]

[0049]
Where Ψ_pyAnd Ψ_pxRepresents a cross-correlation value between the M-sequence signal p (t) and the received signal Y, and a cross-correlation value between the M-sequence signal p (t) and the transmitted signal X, respectively, which are given by the following equations.
[0050]
[Formula 7]

[0051]
[Formula 8]

[0052]
  Here, since the M-sequence signal p (t) and the received signal Y and the M-sequence signal p (t) and the transmitted signal X are not causally related to each other, the respective cross-correlation values Ψ_py, Ψ_pxVaries depending on time t. When the period T of the M-sequence signal p (t) is sufficiently long, the cross-correlation value Ψ_py, Ψ_pxBecomes zero. Therefore, in this case, the second term and the third term on the right side of Equation 6 are zero and can be transformed as the following equation.
[0053]
[Formula 9]

[0054]
Therefore, Ψ_pp(t, t + τ) = Ψ_ppBy calculating a value obtained by normalizing the cross-correlation value of the M-sequence signal (pseudo white signal) p (t) given by (τ) and the response signal q (t) by A, the M-sequence signal p (t ) To the response signal q (t), the coefficient in the delay time τ of the impulse response of the system can be obtained. However, the above equation 9 holds because of the cross-correlation value Ψ given by the equations 7 and 8._py, Ψ_pxIs the case where the period T of the M-sequence signal p (t) is long enough to be regarded as zero. In other cases, the second term and the third term on the right side of the above equation 6 are disturbance components, and the impulse response It becomes a factor which degrades estimation accuracy.
[0055]
  By the way, when the cross-correlation calculating unit 25 is realized by a digital circuit, the above-mentioned Ψ_pp(t, t + τ) = Ψ_ppThe cross-correlation value obtained by normalizing (τ) by A is expressed by the following equation.
[0056]
[Formula 10]

[0057]
Here, I represents the period of the M-sequence signal p (t) in the discrete time system, and p (i) and q (i + j) are the M-sequence signal p (t) and the response signal q at the sample times i and i + j, respectively. Represents the data of (t). Further, the value obtained by the above equation 7 is a coefficient h of the impulse response of the system from the M-sequence signal p (t) to the response signal q (t)._p(j), where the impulse response observation time necessary for estimating the gain margin of the closed-loop system is the sample time J, h_pIt is necessary to obtain a total of J coefficients of (0), h (1), h (2), ..., h (J-1). Therefore, in this embodiment, in order to estimate the impulse response of the system from the M-sequence signal (pseudo white signal) p (t) to the response signal q (t), I × J product-sum operations are required. To do.
[0058]
  The impulse response (response signal Q) of the system from the pseudo white signal p (t) obtained by the arithmetic processing as described above to the response signal q (t) is input to the envelope detector 11 and described in the first embodiment. As described above, the closed loop gain margin estimation unit 12 estimates the gain margin value from the attenuation characteristic with respect to the time axis of the envelope component extracted by the envelope detector 11, and calculates the required insertion loss amount based on the estimated gain margin value. calculate.
[0059]
  As described above, in the present embodiment, a pseudo white signal generator 24 is added to the basic configuration insertion loss amount adjusting means 8 instead of the impulse signal generator 9, and the pseudo white signal and the response to the pseudo white signal are included. A cross-correlation calculation unit 25 for obtaining a cross-correlation value of the signal and estimating a closed-loop impulse response based on the cross-correlation value is added, and the envelope detector 11 and the closed-loop gain margin are estimated from the estimated impulse response (response signal). Since the gain margin value is estimated by the unit 12, the gain margin value can be accurately estimated without giving the user unpleasant discomfort, and the closed-loop system is stabilized by setting the insertion loss amount to an appropriate value. be able to.
[0060]
  (Embodiment 3)
  FIG. 5 is a block diagram showing the insertion loss amount adjusting means 8 in Embodiment 3 of the present invention, in that the synchronization control unit 15 and the synchronization addition averaging unit 16 described in Embodiment 1 are provided for Embodiment 2. There are features.
[0061]
  In this embodiment, when the period of the pseudo white signal transmitted to the signal path is not sufficiently long and the second term and the third term on the right side of Equation 6 are not regarded as zero and become a disturbance element, the synchronous addition average These effects are reduced by processing, and the estimation accuracy of the impulse response of the closed loop system is improved. Now, let the period of the M-sequence signal p (t) be T₁When the number of synchronous additions is N, the output value of the synchronous addition averaging unit 16 is expressed by the following equation.
[0062]
[Formula 11]

[0063]
Substituting Equations 6 and 9 into Equation 11 above and rearranging results in the following equation.
[0064]
[Formula 12]

[0065]
Where the cross-correlation value Ψ_py, Ψ_pxSince the value fluctuates irregularly with respect to time t, the second term and the third term on the right side in the above equation 12 can be suppressed by sufficiently increasing the value of the synchronous addition number N.
[0066]
  According to the present embodiment, the period T of the M-sequence signal p (t) transmitted to the signal path can be shortened as compared with the second embodiment, so that the insertion loss amount adjusting means 8 is realized by a digital circuit. Further, there is an advantage that the capacity of the storage device and the number of product sums required for the generation of the M-sequence signal p (t) and the calculation of Equation 10 can be reduced. The closed loop impulse response (response signal) output from the synchronous addition averaging unit 16 is input to the envelope detector 11 and the required insertion loss amount is calculated by the same arithmetic processing as in the first embodiment..
[0067]
  (Embodiment4)
  FIG.Is an embodiment of the present invention42 is a block diagram showing the insertion loss amount adjusting means 8 in FIG. 1 and is characterized in that a noise level estimation unit 20 and an impulse signal transmission level adjustment unit 21 are provided in the first embodiment.
[0068]
  The noise level estimation unit 20 estimates the level of ambient noise existing in the closed loop before the impulse signal {P} is sent to the closed loop system, and the impulse signal transmission level adjustment unit 21 estimates the ambient noise. The impulse signal P output from the impulse signal generator 9 according to the level of₀Adjusts the level of…. That is, in the gain margin value estimation process according to the present invention, the ratio of the impulse signal transmission level to the ambient noise level in the closed loop (this ratio is defined here as the S / N ratio) is the gain margin value estimation accuracy. The impulse signal P is closed loop.₀The ambient noise level before... Is transmitted is estimated by the noise level estimation unit 20, and the impulse signal transmission level adjustment unit 21 is set so that the S / N ratio becomes a predetermined value or more with respect to the estimated ambient noise level. Impulse signal P₀By adjusting the transmission level of..., The estimation accuracy of the gain margin value can be improved.
[0069]
  As described above, in this embodiment, the noise level estimation unit 20 that estimates the level of ambient noise existing in the closed loop, and the transmission level of the sample signal (impulse signal) are adjusted based on the estimation result by the noise level estimation unit 20. Since the insertion loss amount adjusting unit 8 is provided with the impulse signal transmission level adjusting unit 21 that performs the above operation, the level of the impulse signal transmitted to the closed loop system is set to a sufficient level with respect to the ambient noise level. The response signal {Q} to the impulse signal {P} can be obtained with a high S / N ratio, and the estimation accuracy of the gain margin value can be improved. It should be noted that the same effect can be obtained even if the noise level estimation unit 20 and means for adjusting the transmission level of the pseudo white signal are provided in the configuration of the third embodiment..
[0070]
【The invention's effect】
  According to a first aspect of the present invention, there is provided a microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, and a speaker that rings according to the audio signal on the receiving side. A second amplifying means for amplifying the audio signal output to the speaker, a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side, and the loss inserting means. The control means for variably controlling the amount of loss generated, and the gain margin in the closed loop formed through the microphone and the speaker are estimated according to the response signal to the sample signal sent to the signal path, and the loss based on the estimation result via the control means An insertion loss amount adjusting means for adjusting the amount, the insertion loss amount adjusting means sending a sample signal to the signal path, and a sound An adder for adding the sample signal to the signal, an envelope detector for detecting the envelope of the response signal to the sample signal, and a closed loop gain margin estimating unit for estimating the gain margin in the closed loop based on the output signal of the envelope detector A synchronization control unit that performs timing control for sending the sample signal multiple times from the sample signal generator to the signal path at a predetermined time interval, and timing information from the synchronization control unit that is provided in the preceding stage of the envelope detector And a synchronous addition averaging unit for performing a synchronous addition averaging process on the response signal for the sample signal, so that the acoustic coupling from the speaker to the microphone and between the transmission side and the reception side and the external communication line are It is formed due to reflection due to impedance mismatch in the 2-4 line conversion means used when it is necessary to perform 4 line conversion. Insertion loss by estimating the gain margin value based on the attenuation characteristic of the envelope component of the response signal observed when a sample signal is input to the closed loop system, and calculating the excess or deficiency relative to the desired gain margin value Therefore, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, stable call quality can be maintained without causing howling, and the amount of insertion loss is more than necessary. Therefore, there is an effect that the feasibility of two-way simultaneous call performance is increased. In addition, the synchronization control unit controls the transmission timing of the sample signal, the synchronous addition averaging unit performs the moving average process synchronized with the sample signal, and the envelope detector detects the envelope component from the output signal of the synchronous addition average unit. Therefore, even if ambient noise or audio signals are present in the closed loop, the gain margin can be estimated with sufficient accuracy by reducing these effects, and the insertion loss can be set to an appropriate value. Thus, there is an effect that a stable call can always be realized. The present invention is particularly effective when the ambient noise existing in the closed loop has a wide frequency band and has a small autocorrelation.
[0071]
  According to a second aspect of the present invention, there is provided a microphone that outputs the collected sound as a voice signal on the transmitting side, a first amplifying means that amplifies the voice signal from the microphone, and a speaker that rings according to the voice signal on the receiving side. A second amplifying means for amplifying the audio signal output to the speaker, a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side, and the loss inserting means. The control means for variably controlling the amount of loss generated, and the gain margin in the closed loop formed through the microphone and the speaker are estimated according to the response signal to the sample signal sent to the signal path, and the loss based on the estimation result via the control means An insertion loss amount adjusting means for adjusting the amount, and the insertion loss amount adjusting means inserts a pseudo white signal having a finite period into the signal path. A live device, an adder that adds a pseudo white signal to an audio signal, a cross-correlation value between the pseudo white signal and a response signal for the pseudo white signal, and a closed-loop impulse response is estimated based on the obtained cross-correlation value A cross-correlation calculator, an envelope detector that detects the envelope of the closed-loop impulse response signal estimated by the cross-correlation calculator, and a closed-loop gain that estimates the gain margin in the closed loop based on the output signal of the envelope detector 2-4, which is used when it is necessary to perform 2-4 line conversion between the speaker and the microphone, or between the transmitting side and the receiving side and an external communication line. Observed when a pseudo-white signal with a finite period is input to a closed-loop system formed due to reflection due to impedance mismatch in the line conversion means. The gain margin value is estimated from the attenuation characteristics of the envelope component of the impulse response signal of the closed loop system estimated from the calculated cross-correlation value and the cross-correlation value between the response signal and the pseudo white signal. Since the insertion loss is adjusted by calculating the excess and deficiency with respect to the value, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality can be achieved without causing howling. Since it can be maintained and the amount of insertion loss is not increased more than necessary, there is an effect that the feasibility of bidirectional simultaneous call performance is increased. In addition, since the pseudo white signal is used for the estimation of the gain margin, there is an effect that it is possible to prevent the user from giving unpleasant discomfort.
[0072]
  According to a third aspect of the present invention, a synchronization control unit that performs timing control for sending a pseudo white signal from a pseudo white signal generator to a signal path a plurality of times at a predetermined time interval, and a front stage of an envelope detector are provided. Since the insertion loss amount adjusting means includes a synchronous addition averaging unit that performs synchronous addition averaging processing on the closed-loop impulse response signal estimated by the cross-correlation calculation unit in response to timing information from the synchronization control unit, the synchronization control unit To control the transmission timing of the pseudo white signal, perform the moving average processing synchronized with the pseudo white signal in the synchronous addition averaging unit, and extract the envelope component from the output signal of the synchronous addition averaging unit with the envelope detector , Even when the level of stationary ambient noise existing in the closed loop is large, the effect of ambient noise is reduced and the gain margin value is estimated with sufficient accuracy. Become possible set ON loss appropriate value, there is an effect that it is possible to always achieve stable call. The present invention is particularly effective when the ambient noise existing in the closed loop has a wide frequency band and the autocorrelation is small..
[0073]
  Claim4The invention includes a noise level estimation unit that estimates a level of ambient noise existing in a closed loop, and a transmission level adjustment unit that adjusts the transmission level of a sample signal or a pseudo white signal based on an estimation result by the noise level estimation unit. Since the insertion loss amount adjusting means is provided, by setting the level of the sample signal or pseudo white signal sent to the closed loop system to a sufficient level with respect to the ambient noise level, a high S / N ratio at the observation point. It is possible to obtain a response signal to the sample signal or the pseudo white signal and to improve the estimation accuracy of the gain margin value..
[Brief description of the drawings]
FIG. 1 is a block diagram showing insertion loss amount adjusting means in Embodiment 1. FIG.
FIG. 2 is a block diagram showing a synchronization control unit and a synchronization addition averaging unit in the same as above.
FIG. 3 is an explanatory diagram for explaining the operation described above.
FIG. 4 is a block diagram showing insertion loss amount adjusting means in the second embodiment.
FIG. 5 is a block diagram showing insertion loss amount adjusting means in the third embodiment.
FIG. 6 is a block diagram showing insertion loss amount adjusting means in the fourth embodiment.
[Fig. 7](A) is a block diagram showing the basic configuration of the present invention, (b) is a block diagram showing the basic configuration of the insertion loss amount adjusting means.
[Fig. 8]It is explanatory drawing for demonstrating the closed loop in the same as the above.
[Explanation of symbols]
  8 Insertion loss adjustment means
  9 Impulse signal generator
  10 Adder
  11 Envelope detector
  12 Closed loop gain margin estimation unit
  15 Synchronization control unit
  16 Synchronous addition averaging unit

Claims (4)

  A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion loss that estimates the gain margin in the closed loop formed through the microphone and speaker according to the control signal and the response signal to the sample signal sent to the signal path and adjusts the loss amount based on the estimation result via the control means And an insertion loss amount adjusting means for sending a sample signal to the signal path and a sample signal for the audio signal. An adder for adding the signal, an envelope detector for detecting the envelope of the response signal to the sample signal, a closed loop gain margin estimating unit for estimating the gain margin in the closed loop based on the output signal of the envelope detector, A synchronization control unit that performs timing control for sending a sample signal multiple times from the sample signal generator to the signal path at time intervals, and a sample that receives timing information from the synchronization control unit that is provided in the preceding stage of the envelope detector A loudspeaker having a synchronous addition averaging unit for performing a synchronous addition averaging process on a response signal to a signal.   A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion loss that estimates the gain margin in the closed loop formed through the microphone and speaker according to the control signal and the response signal to the sample signal sent to the signal path and adjusts the loss amount based on the estimation result via the control means A pseudo white signal generator for inserting a pseudo white signal having a finite period in the signal path, and an audio signal. An adder for adding a pseudo-white signal, a cross-correlation calculation unit for obtaining a cross-correlation value of the pseudo-white signal and a response signal for the pseudo-white signal and estimating a closed-loop impulse response based on the obtained cross-correlation value; An envelope detector for detecting the envelope of the closed-loop impulse response signal estimated by the correlation calculation unit, and a closed-loop gain margin estimation unit for estimating the gain margin in the closed loop based on the output signal of the envelope detector. A loudspeaker, characterized by comprising   A synchronization control unit that performs timing control for sending the pseudo white signal to the signal path from the pseudo white signal generator to the signal path at a predetermined time interval, and timing information from the synchronization control unit that is provided in front of the envelope detector 3. The loudspeaker call according to claim 2, wherein the insertion loss amount adjusting means includes a synchronous addition averaging unit for performing synchronous addition averaging processing on the closed-loop impulse response signal estimated by the cross-correlation calculation unit in response to the received signal. Machine. A noise level estimation unit that estimates the level of ambient noise existing in the closed loop and a transmission level adjustment unit that adjusts the transmission level of the sample signal or the pseudo white signal based on the estimation result by the noise level estimation unit adjust the insertion loss amount 4. The loudspeaker as claimed in claim 1, further comprising means .

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