JP3941580B2

JP3941580B2 - Loudspeaker

Info

Publication number: JP3941580B2
Application number: JP2002127557A
Authority: JP
Inventors: 実福島; 博昭竹山; 裕子前田; 章寺澤; 彰洋菊池
Original assignee: Matsushita Electric Works Ltd
Current assignee: Panasonic Electric Works Co Ltd
Priority date: 2002-04-26
Filing date: 2002-04-26
Publication date: 2007-07-04
Anticipated expiration: 2022-04-26
Also published as: JP2003324369A

Description

【０００１】
【発明の属する技術分野】
本発明は、住宅や事務所等で用いられる拡声通話装置に関するものである。
【０００２】
【従来の技術】
従来より、通話時にハンドセットを持つ必要がなく、通話端末から離れた通話者に対して相手側の通話端末から伝送されてくる音声信号をスピーカによって拡声出力し、かつ、上記通話者の発する音声をマイクロホンにより集音して相手側通話端末へ伝送することで拡声通話（ハンズフリー通話）を実現する拡声通話装置が提供されている。このような拡声通話装置においては、通話者が発した音声の一部が相手側通話端末のスピーカからマイクロホンヘの音響結合や通話端末と伝送路との間のインピーダンスの不整合によって生じる反射などが原因で再び受話信号と重畳して帰還することがあり、この帰還成分のレベルが大きい場合には、不快なエコー（音響エコーあるいは回線エコー）として通話者に聴こえてしまう。また、上記音響結合や反射、および自端末における音響結合により通話系に閉ループが形成され、閉ループの一巡利得が１倍を超える周波数成分が存在する場合には、その周波数においてハウリングを生じ、安定した通話を継続することが不可能となる。したがって、通話端末としての拡声通話装置を設計する上で、上述した不快なエコーやハウリングを如何に抑圧するかが重要な課題となる。
【０００３】
このような課題に対して、従来、通話状態（送話状態、受話状態など）を常時推定し、推定結果に基づき適切な配分で送話路および受話路に対して損失を挿入する音声スイッチを用いて閉ループの一巡利得を低減し不快なエコーやハウリングを抑圧する方式が広く用いられてきた。図３は、上記音声スイッチ１０’を備えた拡声通話装置としてのインターホン親機（以下、「親機」と略す）Ｍ’と、相手側通話端末としてのドアホン子器Ｓとからなる、いわゆるハンズフリーインターホンの従来例を示すブロック図である。親機Ｍ’は、マイクロホン１、スピーカ２、２線−４線変換回路３、マイクロホンアンプＧ１、回線（２線の伝送路）への送話信号を増幅する回線出力アンプＧ２、回線からの受話信号を増幅する回線入力アンプＧ３、スピーカアンプＧ４、並びに音声スイッチ１０’で構成される。また、ドアホン子器Ｓはマイクロホン１’、スピーカ２’、２線−４線変換回路３’、マイクロホンアンプＧ１’、並びにスピーカアンプＧ４’で構成される。
【０００４】
音声スイッチ１０’は、送話側の信号経路に損失を挿入する送話側減衰器１１と、受話側の信号経路に損失を挿入する受話側減衰器１２と、送話側及び受話側の各減衰器１１，１２から挿入する損失量を制御する挿入損失量制御部１３’とを具備する。挿入損失量制御部１３’は、受話側減衰器１２の出力点Ｒoutから音響エコー経路Ｈ_ACを介して送話側減衰器１１の入力点Ｔinへ帰還する経路（以下、「音響側帰還経路」という）の音響側帰還利得を推定するとともに、送話側減衰器１１の出力点Ｔoutから回線エコー経路Ｈ_LINを介して受話側減衰器１２の入力点Ｒinへ帰還する経路（以下、「回線側帰還経路」という）の回線側帰還利得を推定し、音響側及び回線側の各帰還利得の推定値に基づいて閉ループに挿入すべき損失量の総和（送話側減衰器１１の挿入損失量と受話側減衰器１２の挿入損失量の和）を算出するとともに、送話信号及び受話信号を監視して通話状態を推定し、この推定結果と上記総和挿入損失量の算出値に応じて送話側減衰器１１及び受話側減衰器１２の各挿入損失量の配分を決定している。
【０００５】
【発明が解決しようとする課題】
ところで、遠端側（相手側通話端末であるドアホン子器Ｓ側）における周囲騒音と、近端側（マイクロホン１側）における周囲騒音とのレベル差が大きい場合、送話信号及び受話信号を監視して通話状態を推定する音声スイッチ１０’の挿入損失量制御部１３’では、例えば遠端側の周囲騒音レベルが大きい状況においては常に受話状態と判定し、近端側の周囲騒音レベルが大きい状況においては常に送話状態と判定してしまい、実際の通話状態に関係なく、通話状態を受話状態又は送話状態の何れか一方に固定してしまう現象（いわゆる音声スイッチ１０’の片倒れ）が生じることがある。
【０００６】
これに対して本発明者らは、受話信号に含まれる遠端側の背景雑音レベルを推定する遠端側背景雑音レベル推定部と、送話信号に含まれる近端側の背景雑音レベルを推定する近端側背景雑音レベル推定部と、挿入損失量制御部１３’で監視している受話信号を増幅する受話偏重モード設定用増幅器と、挿入損失量制御部１３’で監視している送話信号を増幅する送話偏重モード設定用増幅器と、遠端側背景雑音レベル並びに近端側背景雑音レベルの各推定値に応じて受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を調整する偏重モード制御部とを設け、この偏重モード制御部により、遠端側背景雑音レベルの推定値が近端側背景雑音レベルの推定値よりも充分に大きい値であれば送話偏重モード設定用増幅器の利得を受話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を送話偏重モードに設定し、近端側背景雑音レベルの推定値が遠端側背景雑音レベルの推定値よりも充分に大きい値であれば受話偏重モード設定用増幅器の利得を送話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を受話偏重モードに設定し、遠端側背景雑音レベルの推定値と近端側背景雑音レベルの推定値の差が充分に大きい値でなければ受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を略ゼロとして通話処理手段を中立モードに設定することによって、遠端側の通話端末における周囲騒音のレベルと近端側の拡声通話装置（親機Ｍ’）における周囲騒音のレベルとの差が大きい場合でも周囲騒音のレベルが高い方に通話方向が片倒れすることを防止したものを既に出願している（特願２００１−１７５５２１参照）。
【０００７】
しかしながら、本発明者らによる上記拡声通話装置において、例えば送話偏重モードのときには送話偏重モード設定用増幅器の利得を増大させているために所謂受話ブロッキングが生じ易くなり、あるいは受話偏重モードのときには受話偏重モード設定用増幅器の利得を増大させているために所謂送話ブロッキングが生じ易くなってしまう。ここで受話ブロッキングとは、近端側が無音の状態で遠端側より音声が入力されたときに、近端側のスピーカ−マイクロホン間の音響結合によって生じる音響エコー信号の送話偏重モード設定用増幅器の出力点におけるパワーレベルが、原信号（受話信号）の受話偏重モード設定用増幅器の出力点におけるパワーレベルよりも大きくなって音声スイッチが送話状態になってしまうことにより、遠端側から入力された音声が近端側で受聴できなくなる現象を言う。また送話ブロッキングとは、遠端側が無音の状態で近端側より音声が入力されたときに、遠端側における音響結合又は伝送処理手段における信号の回り込みによって生じる回線エコー信号の受話偏重モード設定用増幅器の出力点におけるパワーレベルが、原信号（送話信号）の送話偏重モード設定用増幅器の出力点におけるパワーレベルよりも大きくなって音声スイッチが受話状態になってしまうことにより、近端側から入力された音声が遠端側で受聴できなくなる現象を言う。
【０００８】
本発明は上記事情に鑑みて為されたものであり、その目的は、送話ブロッキング及び受話ブロッキングを生じることなく、音声スイッチの片倒れを防止した拡声通話装置を提供することにある。
【０００９】
【課題を解決するための手段】
請求項１の発明は、上記目的を達成するために、集音した音声を送話信号として出力するマイクロホンと、相手側の通話端末からの受話信号に応じて鳴動するスピーカと、ハウリングを抑制して拡声通話を可能とする通話処理手段とを備え、通話処理手段は、マイクロホンとスピーカの音響結合によって音響エコーが生じる音響エコー経路、並びに相手側の通話端末における音響結合又は伝送処理手段における信号の回り込みによって回線エコーが生じる回線エコー経路により形成される閉ループの一巡利得を低減してハウリングを抑制する音声スイッチとを有し、音声スイッチは、送話側の信号経路に損失を挿入する送話側損失挿入手段と、受話側の信号経路に損失を挿入する受話側損失挿入手段と、送話信号と受話信号のレベルを比較して送話側及び受話側の各損失挿入手段から挿入する損失量を制御する挿入損失量制御手段とを具備する拡声通話装置であって、通話処理手段は、受話信号に含まれる遠端側の背景雑音レベルを推定する遠端側背景雑音レベル推定部と、送話信号に含まれる近端側の背景雑音レベルを推定する近端側背景雑音レベル推定部と、挿入損失量制御手段で監視している受話信号を増幅する受話偏重モード設定用増幅器と、挿入損失量制御手段で監視している送話信号を増幅する送話偏重モード設定用増幅器と、受話側損失挿入手段の出力点から音響エコー経路を介して送話側損失挿入手段の入力点へ帰還する経路の音響側帰還利得を推定する音響側帰還利得推定部と、送話側損失挿入手段の出力点から回線エコー経路を介して受話側損失挿入手段の入力点へ帰還する経路の回線側帰還利得を推定する回線側帰還利得推定部と、遠端側背景雑音レベル、近端側背景雑音レベル、音響側帰還利得並びに回線側帰還利得の各推定値に応じて受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を調整する偏重モード制御部とを備え、偏重モード制御部は、遠端側背景雑音レベルの推定値が近端側背景雑音レベルの推定値よりも充分に大きい値であり且つ音響側帰還利得の推定値が所定値よりも小さい値となる状態が一定時間以上継続すれば送話偏重モード設定用増幅器の利得を受話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を送話偏重モードに設定し、近端側背景雑音レベルの推定値が遠端側背景雑音レベルの推定値よりも充分に大きい値であり且つ回線側帰還利得の推定値が所定値よりも小さい値となる状態が一定時間以上継続すれば受話偏重モード設定用増幅器の利得を送話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を受話偏重モードに設定し、遠端側背景雑音レベルの推定値と近端側背景雑音レベルの推定値の差の絶対値が充分に大きい値となる状態が一定時間以上継続しなければ受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器で受話信号及び送話信号を増幅しないことにより通話処理手段を中立モードに設定することを特徴とし、遠端側と近端側の周囲騒音のレベル差だけでなく、音響側帰還利得並びに回線側帰還利得の各推定値も考慮して送話偏重モード、受話偏重モード、中立モードに設定しているため、送話ブロッキング及び受話ブロッキングを生じることなく、音声スイッチの片倒れを防止することができる。
【００１０】
請求項２の発明は、請求項１の発明において、偏重モード制御部は、受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を音響側帰還利得並びに回線側帰還利得の推定値に応じて段階的に増減させることを特徴とし、音声スイッチを受話状態あるいは送話状態に切り換えるために必要な遠端側又は近端側の発声音圧レベルを下げることができる。
【００１１】
請求項３の発明は、請求項１の発明において、偏重モード制御部は、受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を遠端側背景雑音レベル並びに近端側背景雑音レベルの推定値に応じて段階的に増減させることを特徴とし、音声スイッチを送話状態と受話状態にバランスよく切り換えることができる。
【００１２】
【発明の実施の形態】
図１は本実施形態の拡声通話装置を示すブロック図である。なお、本実施形態は従来例で説明したハンズフリーインターホンの親機Ｍとして拡声通話装置を構成したものであり、また、本実施形態の基本構成は本発明者らがした先の出願（特願２００１−１７５５２１）に記載した実施形態と共通である。なお、図１では相手側通話端末であるドアホン子機Ｓの図示を省略している。
【００１３】
本実施形態は、マイクロホン１、スピーカ２、２線−４線変換回路３、マイクロホンアンプＧ１、回線（２線の伝送路）への送話信号を増幅する回線出力アンプＧ２、回線からの受話信号を増幅する回線入力アンプＧ３、スピーカアンプＧ４、送話音量調整用増幅器Ｇ５、受話音量調整用増幅器Ｇ６、第１及び第２のエコーキャンセラ３０Ａ，３０Ｂ、並びに音声スイッチ１０で構成される。
【００１４】
第１のエコーキャンセラ３０Ａは適応フィルタ３１Ａと減算器３２Ａからなる従来周知の構成を有し、スピーカ２−マイクロホン１間の音響結合により形成される音響エコー経路Ｈ_ACのインパルス応答を適応フィルタ３１Ａにより適応的に同定し、参照信号（スピーカアンプＧ４への入力信号）から推定したエコー成分（音響エコー）を減算器３２ＡによりマイクロホンアンプＧ１の出力信号（図１における点Ａの信号）から減算することでエコー成分を相殺して消去するものである。また、第２のエコーキャンセラ３０Ｂも適応フィルタ３１Ｂと減算器３２Ｂからなる従来周知の構成を有し、２線−４線変換回路３と伝送路との間のインピーダンスの不整合による反射およびドアホン子機Ｓにおけるスピーカ２’−マイクロホン１’間の音響結合とにより形成される回線エコー経路Ｈ_LINのインパルス応答を適応フィルタ３１Ｂにより適応的に同定し、参照信号（回線出力アンプＧ２への入力信号、すなわち送話信号）から推定したエコー成分（回線エコー）を減算器３２Ｂにより受話信号（図１における点Ｃの信号）から減算することでエコー成分を相殺して消去するものである。
【００１５】
音声スイッチ１０は、送話側の信号経路に損失を挿入する送話側損失挿入手段たる送話側減衰器１１と、受話側の信号経路に損失を挿入する受話側損失挿入手段たる受話側減衰器１２と、送話側及び受話側の各減衰器１１，１２から挿入する損失量を制御する挿入損失量制御部１３とを具備する。挿入損失量制御部１３は、受話側減衰器１２の出力点Ｒoutから音響エコー経路Ｈ_ACを介して送話側減衰器１１の入力点Ｔinへ帰還する経路（以下、「音響側帰還経路」という）の音響側帰還利得αを推定するとともに、送話側減衰器１１の出力点Ｔoutから回線エコー経路Ｈ_LINを介して受話側減衰器１２の入力点Ｒinへ帰還する経路（以下、「回線側帰還経路」という）の回線側帰還利得βを推定し、音響側及び回線側の各帰還利得α，βの推定値α’，β’に基づいて閉ループに挿入すべき損失量の総和（送話側減衰器１１の挿入損失量と受話側減衰器１２の挿入損失量の和）を算出する総損失量算出部１４と、送話信号及び受話信号を監視して通話状態を推定し、この推定結果と総損失量算出部１４の算出値に応じて送話側減衰器１１及び受話側減衰器１２の各挿入損失量の配分を決定する挿入損失量分配処理部１５とからなる。なお、第１及び第２のエコーキャンセラ３０Ａ，３０Ｂと音声スイッチ１０を含む通話処理手段は、ＤＳＰ（Digital Signal Processor）を用いて従来周知の技術により実現可能である。
【００１６】
総損失量算出部１４では、整流平滑器や低域通過フィルタ等を用いて送話側減衰器１１の入力信号の短時間における時間平均パワーを推定し、同じく整流平滑器や低域通過フィルタ等を用いて受話側減衰器１２の出力信号の短時間における時間平均パワーを推定し、音響側帰還経路Ｈ_ACにて想定される最大遅延時間において受話側減衰器１２の出力信号の時間平均パワーの推定値の最小値を求め、この最小値で送話側減衰器１１の入力信号の時間平均パワーの推定値を除算した値を音響側帰還利得αの推定値α’とするとともに、整流平滑器や低域通過フィルタ等を用いて受話側減衰器１２の入力信号の短時間における時間平均パワーを推定し、同じく整流平滑器や低域通過フィルタ等を用いて送話側減衰器１１の出力信号の短時間における時間平均パワーを推定し、回線側帰還経路Ｈ_LINにて想定される最大遅延時間において送話側減衰器１１の出力信号の時間平均パワーの推定値の最小値を求め、この最小値で受話側減衰器１２の入力信号の時間平均パワーの推定値を除算した値を回線側帰還利得βの推定値β’とする。そして、総損失量算出部１４は音響側帰還利得α及び回線側帰還利得βの各推定値α’，β’から所望の利得余裕ＭＧを得るために必要な総損失量Ｌｔを算出し、その値Ｌｔを挿入損失量分配処理部１５に出力する。
【００１７】
挿入損失量分配処理部１５では、送話側減衰器１１の入出力信号及び受話側減衰器１２の入出力信号を監視し、これらの信号のパワーレベルの大小関係並びに音声信号の有無などの情報から通話状態（受話状態、送話状態等）を判定するとともに、判定された通話状態に応じた割合で総損失量Ｌｔを送話側減衰器１１と受話側減衰器１２に分配するように各減衰器１１，１２の挿入損失量を調整する。
【００１８】
ところで本実施形態における総損失量算出部１４は、上述のように各帰還利得α，βの推定値α’，β’に基づいて閉ループに挿入すべき損失量の総和を算出して適応更新する更新モード、並びに総損失量を所定の初期値に固定する固定モードの２つの動作モードを有し、相手側通話端末（ドアホン子機Ｓ）との通話開始から第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが充分に収束するまでの期間には固定モードで動作するとともに第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが充分に収束した後の期間には更新モードで動作する。すなわち、総損失量算出部１４では音響側帰還利得α及び回線側帰還利得βの推定値α’，β’がともに通話開始から所定時間（数百ミリ秒）以上継続して所定の閾値ε（例えば、通話開始時における各推定値α’，β’に対して１０ｄＢ〜１５ｄＢ小さい値）を下回った時点で第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが充分に収束したものとみなし、上記時点以前には総損失量を初期値に固定する固定モードで動作し、上記時点以降には各推定値α’，β’に基づいて総損失量を適応更新する更新モードに動作モードを切り換える。なお、固定モードにおける総損失量の初期値は更新モードにおいて随時更新される総損失量よりも充分に大きな値に設定される。
【００１９】
而して、通話開始直後の第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが充分に収束していない状態においては、固定モードで動作する総損失量算出部１４によって充分に大きな値に設定される初期値の総損失量が閉ループに挿入されるため、不快なエコー（音響エコー並びに回線エコー）やハウリングの発生を抑制して安定した半二重通話を実現することができる。また、通話開始から時間が経過して第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが充分に収束した状態においては、総損失量算出部１４の動作モードが固定モードから更新モードに切り換わって閉ループに挿入する総損失量が初期値よりも充分に低い値に減少するため、双方向の同時通話が実現できるものである。しかも、総損失量の初期値を適切な値に設定することにより、通話開始直後の第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが収束していない状態のハウリング防止のために閉ループの一巡利得が１倍を超えないように各増幅器の利得を設計するという制約がなくなり、親機Ｍのハウジング（図示せず）の形状や構造等に関わらずに所望の通話音量が得られるように増幅器の利得を設計することができる。
【００２０】
次に、更新モードにおける総損失量算出部１４の具体的な動作を図２のフローチャートを参照して説明する。
【００２１】
総損失量算出部１４は、固定モードから更新モードに移行した時点（ｔ＝ｔ１）から所定のサンプリング周期で音響側帰還利得α並びに回線側帰還利得βの推定処理を実行してその推定値α'(n)，β'(n)を算出し（ステップ１）、これら２つの推定値α'(n)，β'(n)の積と利得余裕ＭＧとから、閉ループの利得余裕をＭＧ[ｄＢ]に保つために必要とされる総損失量所望値Ｌｒ(n)を下式により算出する（ステップ２）。
【００２２】
Ｌｒ(n)＝２０log|α'(n)・β'(n)|＋ＭＧ[ｄＢ]
なお、α'(n)，β'(n)，Ｌｒ(n)はそれぞれ更新モード移行時点からｎ回目のサンプリングによって算出された帰還利得の推定値並びに総損失量所望値を示す。さらに、総損失量算出部１４は上式から算出したｎ回目の総損失量所望値Ｌｒ(n)と、前回（ｎ−１回目）の総損失量Ｌｔ(n-1)、すなわち前回の処理で決定されて実際に挿入された総損失量に対して今回算出した総損失量所望値Ｌｒ(n)が大きい場合、前回の総損失量Ｌｔ(n-1)に微少な増加量Δｉ[ｄＢ]を加算した値を今回の総損失量Ｌｔ(n)＝Ｌｔ(n-1)＋Δｉとし（ステップ３、ステップ４）、前回の総損失量Ｌｔ(n-1)に対して今回算出した総損失量所望値Ｌｒ(n)が小さい場合、前回の総損失量Ｌｔ(n-1)から微少な減少量Δｄ[ｄＢ]を減算した値を今回の総損失量Ｌｔ(n)＝Ｌｔ(n-1)−Δｄとする（ステップ５、ステップ６）。
【００２３】
このように総損失量算出部１４による総損失量の増減をΔｉ又はΔｄの微少な値に抑えることにより、相手側通話端末（ドアホン子機Ｓ）との通話開始直後のように第１及び第２のエコーキャンセラ３０Ａ，３０Ｂが収束に向かって活発に係数を更新しているために音響側帰還利得α及び回線側帰還利得βの変化が激しい状態においても、聴感上の違和感をなくすことができる。
【００２４】
一方、本実施形態の要旨は、受話信号に含まれる遠端側（相手側の通話端末）の背景雑音レベルを推定する遠端側背景雑音レベル推定部２０と、送話信号に含まれる近端側（マイクロホン１側）の背景雑音レベルを推定する近端側背景雑音レベル推定部２１と、挿入損失量制御部１３で監視している受話信号を増幅する受話偏重モード設定用増幅器ＧＲと、挿入損失量制御部１３で監視している送話信号を増幅する送話偏重モード設定用増幅器ＧＴと、受話側減衰器１２の出力点Ｒoutから音響エコー経路Ｈ_ACを介して送話側減衰器１１の入力点Ｔinへ帰還する経路の音響側帰還利得αを推定する音響側帰還利得推定部２２と、送話側減衰器１１の出力点Ｔoutから回線エコー経路Ｈ_LINを介して受話側減衰器１２の入力点Ｒinへ帰還する経路の回線側帰還利得βを推定する回線側帰還利得推定部２３と、遠端側背景雑音レベル、近端側背景雑音レベル、音響側帰還利得並びに回線側帰還利得の各推定値に応じて受話偏重モード設定用増幅器ＧＲ並びに送話偏重モード設定用増幅器ＧＴの各利得を調整する偏重モード制御部２４とを、第１及び第２のエコーキャンセラ３０Ａ，３０Ｂ並びに音声スイッチ１０を含む通話処理手段に備えた点にある。なお、遠端側背景雑音レベル推定部２０、近端側背景雑音レベル推定部２１、音響側帰還利得推定部２２、回線側帰還利得推定部２３並びに偏重モード制御部２４は第１及び第２のエコーキャンセラ３０Ａ，３０Ｂや音声スイッチ１０とともにＤＳＰを用いて実現される。
【００２５】
遠端側背景雑音レベル推定部２０並びに近端側背景雑音レベル推定部２１は何れも、立ち上がりが緩やかであり且つ立ち下がりが急峻な特性をもつ積分回路又はデジタルフィルタ等によって実現され、遠端側背景雑音レベル推定部２０では受話音量調整用増幅器Ｇ６から出力される受話信号中に定常的に存在する暗騒音（背景雑音）レベルを推定し、近端側背景雑音レベル推定部２１では送話音量調整用増幅器Ｇ５から出力される送話信号中に定常的に存在する背景雑音レベルを推定する。なお、音響側帰還利得推定部２２並びに回線側帰還利得推定部２３における各帰還利得の推定処理は総損失量算出部１４による推定処理と共通である。
【００２６】
偏重モード制御部２４は、遠端側背景雑音レベルの推定値ＰＦｎが近端側背景雑音レベルの推定値ＰＮｎよりも充分に大きい値（ＰＦｎ≫ＰＮｎ）であり且つ音響側帰還利得αの推定値｜α'｜が所定値α０よりも小さい値（｜α'｜＜α０）となる状態が一定時間Ｔ１［秒］以上継続すれば、送話偏重モード設定用増幅器ＧＴの利得をＧ１［ｄＢ］（Ｇ１＞０）、受話偏重モード設定用増幅器ＧＲの利得を０［ｄＢ］とすることで音声スイッチ１０を送話偏重モードに設定し、近端側背景雑音レベルの推定値ＰＮｎが遠端側背景雑音レベルの推定値ＰＦｎよりも充分に大きい値（ＰＮｎ≫ＰＦｎ）であり且つ回線側帰還利得βの推定値｜β'｜が所定値β０よりも小さい値（｜β'｜＜β０）となる状態が一定時間Ｔ２［秒］以上継続すれば、受話偏重モード設定用増幅器ＧＲの利得をＧ２［ｄＢ］（Ｇ２＞０）、送話偏重モード設定用増幅器ＧＴの利得を０［ｄＢ］とすることで音声スイッチ１０を受話偏重モードに設定し、遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎの差が充分に大きい値となる状態が一定時間Ｔ１又はＴ２以上継続しなければ、受話偏重モード設定用増幅器ＧＲ並びに送話偏重モード設定用増幅器ＧＴで受話信号及び送話信号を増幅しない、つまり各々の利得を略ゼロとすることにより音声スイッチ１０を中立モードに設定する。
【００２７】
すなわち、遠端側の周囲騒音レベルと近端側の周囲騒音レベルとの差が大きい場合、送話信号及び受話信号を監視して通話状態を推定する挿入損失量制御部１３では、例えば遠端側の周囲騒音レベルが大きい状況においては常に受話状態と判定し、近端側の周囲騒音レベルが大きい状況においては常に送話状態と判定してしまい、実際の通話状態に関係なく、受話状態又は送話状態の何れか一方に通話状態を固定してしまう現象（所謂音声スイッチ１０の片倒れ）が生じることがある。これに対して本発明者らは、偏重モード制御部２４が遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎを比較し、遠端側背景雑音レベルの推定値ＰＦｎの方が充分に大きい場合は挿入損失量制御部１３で監視する送話信号を送話偏重モード設定用増幅器ＧＴで利得Ｇ１［ｄＢ］だけ増幅することによって挿入損失量制御部１３が送話状態と判定し易い状態（送話偏重モード）に設定し、反対に近端側背景雑音レベルの推定値ＰＮｎの方が充分に大きい場合は挿入損失量制御部１３で監視する受話信号を受話偏重モード設定用増幅器ＧＲで利得Ｇ２［ｄＢ］だけ増幅することによって挿入損失量制御部１３が受話状態と判定し易い状態（受話偏重モード）に設定するようにして音声スイッチ１０の片倒れを防止した拡声通話装置を既に提案している。しかしながら、偏重モード制御部２４が遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎの比較結果のみに基づいて送話偏重モード、受話偏重モード並びに中立モードの切り換えを行うと、例えば音響側帰還利得αがある程度大きな値である場合に、本来ならば受話状態となるべき状況であるのに受話状態とならない現象（受話ブロッキング）や、回線側帰還利得βがある程度大きな値である場合に、本来ならば送話状態となるべき状況であるのに送話状態とならない現象（送話ブロッキング）が生じる虞がある。
【００２８】
そこで本発明においては、偏重モード制御部２４が遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎのレベル差だけでなく、音響側帰還利得α並びに回線側帰還利得βの各推定値｜α'｜，｜β'｜も考慮して送話偏重モード、受話偏重モード、中立モードに設定している。すなわち、遠端側背景雑音レベルの推定値ＰＦｎが近端側背景雑音レベルの推定値ＰＮｎよりも充分に大きい状況が一定時間Ｔ１以上継続した場合でも、音響側帰還利得αの推定値｜α'｜が所定値α０を上回るときには送話偏重モードに移行しないため、受話ブロッキングを生じることなく、音声スイッチ１０の片倒れを防止することができる。また、近端側背景雑音レベルの推定値ＰＮｎが遠端側背景雑音レベルの推定値ＰＦｎよりも充分に大きい状況が一定時間Ｔ２以上継続した場合でも、回線側帰還利得βの推定値｜β'｜が所定値β０を上回るときには受話偏重モードに移行しないため、送話ブロッキングを生じることなく、音声スイッチ１０の片倒れを防止することができる。
【００２９】
ここで、受話ブロッキング及び送話ブロッキングを防止するためには、｜α'｜＋Ｇ１＜Ｍ１，｜β'｜＋Ｇ２＜Ｍ２（これを書き換えると、｜α'｜＜Ｍ１−Ｇ１＝α０，｜β'｜＜Ｍ２−Ｇ２＝β０）の関係が成立していなければならない。すなわち、上記所定値α０，β０は受話ブロッキング並びに送話ブロッキングに対するマージンＭ１，Ｍ２と送話偏重モード設定用増幅器ＧＴの利得Ｇ１及び受話偏重モード設定用増幅器ＧＲの利得Ｇ２に応じて決定される。なお、送話偏重モード設定用増幅器ＧＴの利得Ｇ１並びに受話偏重モード設定用増幅器ＧＲの利得Ｇ２は、それぞれ使用環境において想定される遠端側及び近端側の背景雑音レベルに対して、近端側及び遠端側からどの程度の音圧レベルで発声したときに音声スイッチ１０を送話状態あるいは受話状態に切り換えるかという拡声通話装置としての要求仕様に基づいて決定される。
【００３０】
ところで、偏重モード制御部２４が中立モードから送話偏重モードあるいは受話偏重モードに切り換える場合に、送話偏重モード設定用増幅器ＧＴ並びに受話偏重モード設定用増幅器ＧＲの各利得Ｇ１，Ｇ２を音響側帰還利得α並びに回線側帰還利得βの推定値｜α'｜，｜β'｜に応じて段階的に増減させるようにしてもよい。例えば、利得Ｇ１を３段階に増減させるとすると、偏重モード制御部２４は、音響側帰還利得αの推定値｜α'｜が｜α'｜≧α０₁のときにＧＴ＝０、α０₂≦｜α'｜＜α０₁のときにＧＴ＝Ｇ１”、α０₃≦｜α'｜＜α０₂のときにＧＴ＝Ｇ１’、｜α'｜＜α０₃のときにＧＴ＝Ｇ１となるように送話偏重モード設定用増幅器ＧＴの利得Ｇ１を増減する（但し、α０₃＜α０₂＜α０₁、０＜Ｇ１”＜Ｇ１’＜Ｇ１）。同じく、利得Ｇ２を３段階に増減させるとすると、偏重モード制御部２４は、回線側帰還利得βの推定値｜β'｜が｜β'｜≧β０₁のときにＧＲ＝０、β０₂≦｜β'｜＜β０₁のときにＧＲ＝Ｇ２”、β０₃≦｜β'｜＜β０₂のときにＧＲ＝Ｇ２’、｜β'｜＜β０₃のときにＧＲ＝Ｇ２となるように受話偏重モード設定用増幅器ＧＲの利得Ｇ２を増減する（但し、β０₃＜β０₂＜β０₁、０＜Ｇ２”＜Ｇ２’＜Ｇ２）。
【００３１】
例えば、送話偏重モード設定用増幅器ＧＴの利得Ｇ１が大きな値に設定される場合には所定値α０が小さい値に設定されることになり、音響側帰還利得αの推定値｜α'｜の初期値が大きい場合や第１の音響エコーキャンセラ３０Ａの収束速度が遅い場合には、推定値｜α'｜が所定値α０を下回るまでに要する時間が長くなり、状況によっては通話中に推定値｜α'｜が所定値α０を下回る状態に達しないこともある。このような場合においても、上述のように偏重モード制御部２４が中立モードから送話偏重モードに切り換える際に、送話偏重モード設定用増幅器ＧＴの利得Ｇ１を音響側帰還利得αの推定値｜α'｜に応じて段階的に増減させれば、音声スイッチ１０を送話状態に切り換えるために必要な近端側の発声音圧レベルを下げることができ、音声スイッチ１０が送話状態に切り換わり易くなる。同様に、受話偏重モード設定用増幅器ＧＲの利得Ｇ２が大きな値に設定される場合には所定値β０が小さい値に設定されることになり、回線側帰還利得βの推定値｜β'｜の初期値が大きい場合や第２の音響エコーキャンセラ３０Ｂの収束速度が遅い場合には、推定値｜β'｜が所定値β０を下回るまでに要する時間が長くなり、状況によっては通話中に推定値｜β'｜が所定値β０を下回る状態に達しないこともある。このような場合においても、上述のように偏重モード制御部２４が中立モードから受話偏重モードに切り換える際に、受話偏重モード設定用増幅器ＧＲの利得Ｇ２を回線側帰還利得βの推定値｜β'｜に応じて段階的に増減させれば、音声スイッチ１０を受話状態に切り換えるために必要な遠端側の発声音圧レベルを下げることができ、音声スイッチ１０が受話状態に切り換わり易くなる。
【００３２】
また、偏重モード制御部２４が中立モードから送話偏重モードあるいは受話偏重モードに切り換える場合に、送話偏重モード設定用増幅器ＧＴ並びに受話偏重モード設定用増幅器ＧＲの各利得Ｇ１，Ｇ２を遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎとの差に応じて段階的に増減させるようにしてもよい。例えば、利得Ｇ１を３段階に増減させるとすると、偏重モード制御部２４は、上記差Ｘ（＝ＰＦｎ−ＰＮｎ）がＸ≦Ｘ１のときにＧＴ＝０、Ｘ１＜Ｘ≦Ｘ２のときにＧＴ＝Ｇ１”、Ｘ２＜Ｘ≦Ｘ３のときにＧＴ＝Ｇ１’、Ｘ＞Ｘ３のときにＧＴ＝Ｇ１となるように送話偏重モード設定用増幅器ＧＴの利得Ｇ１を増減する（但し、Ｘ１＜Ｘ２＜Ｘ３、０＜Ｇ１”＜Ｇ１’＜Ｇ１）。同じく、利得Ｇ２を３段階に増減させるとすると、偏重モード制御部２４は、上記差Ｙ（＝ＰＮｎ−ＰＦｎ）がＹ≦Ｙ１のときにＧＲ＝０、Ｙ１＜Ｙ≦Ｙ２のときにＧＲ＝Ｇ２”、Ｙ２＜Ｙ≦Ｙ３のときにＧＲ＝Ｇ２’、Ｙ＞Ｙ３のときにＧＲ＝Ｇ２となるように受話偏重モード設定用増幅器ＧＲの利得Ｇ２を増減する（但し、Ｙ１＜Ｙ２＜Ｙ３、０＜Ｇ２”＜Ｇ２’＜Ｇ２）。
【００３３】
上述のように偏重モード制御部２４が中立モードから送話偏重モードあるいは受話偏重モードに切り換える場合に、送話偏重モード設定用増幅器ＧＴ並びに受話偏重モード設定用増幅器ＧＲの各利得Ｇ１，Ｇ２を遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎとの差に応じて段階的に増減させるようにすれば、例えば遠端側背景雑音レベルの推定値ＰＦｎと近端側背景雑音レベルの推定値ＰＮｎとの差が大きく、且つ利得Ｇ１，Ｇ２の設定値が大きい場合、近端側及び遠端側の背景雑音レベル差に応じて音声スイッチ１０を送話状態と受話状態にバランスよく切り換えることができる。
【００３４】
【発明の効果】
請求項１の発明は、集音した音声を送話信号として出力するマイクロホンと、相手側の通話端末からの受話信号に応じて鳴動するスピーカと、ハウリングを抑制して拡声通話を可能とする通話処理手段とを備え、通話処理手段は、マイクロホンとスピーカの音響結合によって音響エコーが生じる音響エコー経路、並びに相手側の通話端末における音響結合又は伝送処理手段における信号の回り込みによって回線エコーが生じる回線エコー経路により形成される閉ループの一巡利得を低減してハウリングを抑制する音声スイッチとを有し、音声スイッチは、送話側の信号経路に損失を挿入する送話側損失挿入手段と、受話側の信号経路に損失を挿入する受話側損失挿入手段と、送話信号と受話信号のレベルを比較して送話側及び受話側の各損失挿入手段から挿入する損失量を制御する挿入損失量制御手段とを具備する拡声通話装置であって、通話処理手段は、受話信号に含まれる遠端側の背景雑音レベルを推定する遠端側背景雑音レベル推定部と、送話信号に含まれる近端側の背景雑音レベルを推定する近端側背景雑音レベル推定部と、挿入損失量制御手段で監視している受話信号を増幅する受話偏重モード設定用増幅器と、挿入損失量制御手段で監視している送話信号を増幅する送話偏重モード設定用増幅器と、受話側損失挿入手段の出力点から音響エコー経路を介して送話側損失挿入手段の入力点へ帰還する経路の音響側帰還利得を推定する音響側帰還利得推定部と、送話側損失挿入手段の出力点から回線エコー経路を介して受話側損失挿入手段の入力点へ帰還する経路の回線側帰還利得を推定する回線側帰還利得推定部と、遠端側背景雑音レベル、近端側背景雑音レベル、音響側帰還利得並びに回線側帰還利得の各推定値に応じて受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を調整する偏重モード制御部とを備え、偏重モード制御部は、遠端側背景雑音レベルの推定値が近端側背景雑音レベルの推定値よりも充分に大きい値であり且つ音響側帰還利得の推定値が所定値よりも小さい値となる状態が一定時間以上継続すれば送話偏重モード設定用増幅器の利得を受話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を送話偏重モードに設定し、近端側背景雑音レベルの推定値が遠端側背景雑音レベルの推定値よりも充分に大きい値であり且つ回線側帰還利得の推定値が所定値よりも小さい値となる状態が一定時間以上継続すれば受話偏重モード設定用増幅器の利得を送話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を受話偏重モードに設定し、遠端側背景雑音レベルの推定値と近端側背景雑音レベルの推定値の差の絶対値が充分に大きい値となる状態が一定時間以上継続しなければ受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器で受話信号及び送話信号を増幅しないことにより通話処理手段を中立モードに設定するので、遠端側と近端側の周囲騒音のレベル差だけでなく、音響側帰還利得並びに回線側帰還利得の各推定値も考慮して送話偏重モード、受話偏重モード、中立モードに設定しているため、送話ブロッキング及び受話ブロッキングを生じることなく、音声スイッチの片倒れを防止することができるという効果がある。
【００３５】
請求項２の発明は、請求項１の発明において、偏重モード制御部は、受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を音響側帰還利得並びに回線側帰還利得の推定値に応じて段階的に増減させるので、音声スイッチを受話状態あるいは送話状態に切り換えるために必要な遠端側又は近端側の発声音圧レベルを下げることができるという効果がある。
【００３６】
請求項３の発明は、請求項１の発明において、偏重モード制御部は、受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を遠端側背景雑音レベル並びに近端側背景雑音レベルの推定値に応じて段階的に増減させるので、音声スイッチを送話状態と受話状態にバランスよく切り換え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】実施形態を示すブロック図である。
【図２】同上の動作説明用のフローチャートである。
【図３】従来例を示すブロック図である。
【符号の説明】
１マイクロホン
２スピーカ
１０音声スイッチ
１３挿入損失量制御部
２０遠端側背景雑音レベル推定部
２１近端側背景雑音レベル推定部
２２音響側帰還利得推定部
２３回線側帰還利得推定部
２４偏重モード制御部[0001]
BACKGROUND OF THE INVENTION
The present invention relates to a loudspeaker device used in a house, an office, or the like.
[0002]
[Prior art]
Conventionally, it is not necessary to have a handset during a call, and a voice signal transmitted from the other party's call terminal is output to the caller away from the call terminal using a speaker, and the voice emitted by the caller is output. 2. Description of the Related Art There is provided a loudspeaker device that implements a loudspeaker call (hands-free call) by collecting sound with a microphone and transmitting the collected sound to a caller terminal. In such a loudspeaker, a part of the voice uttered by the caller is reflected by acoustic coupling from the speaker of the other party's call terminal to the microphone or impedance mismatch between the call terminal and the transmission line. For this reason, there may be a case where feedback is again superimposed on the received signal, and if the level of the feedback component is high, the caller hears it as an unpleasant echo (acoustic echo or line echo). In addition, when a closed loop is formed in the communication system due to the above acoustic coupling and reflection, and acoustic coupling at the terminal itself, and there is a frequency component in which the loop gain of the closed loop exceeds one time, howling occurs at that frequency, and stable It becomes impossible to continue the call. Therefore, how to suppress the above-mentioned unpleasant echo and howling is an important issue in designing a loudspeaker device as a call terminal.
[0003]
Conventionally, a voice switch that always estimates the call state (transmission state, reception state, etc.) and inserts losses into the transmission path and reception path with appropriate distribution based on the estimation results. A method of reducing closed loop loop gain and suppressing unpleasant echoes and howling has been widely used. FIG. 3 shows a so-called hands composed of an interphone master unit (hereinafter abbreviated as “master unit”) M ′ as a loudspeaker device provided with the voice switch 10 ′ and a doorphone slave unit S as a counterpart call terminal. It is a block diagram which shows the prior art example of a free intercom. Base unit M ′ includes microphone 1, speaker 2, two-wire / four-wire conversion circuit 3, microphone amplifier G 1, line output amplifier G 2 that amplifies a transmission signal to the line (two-wire transmission line), and reception from the line. A line input amplifier G3 for amplifying the signal, a speaker amplifier G4, and a voice switch 10 ′ are included. Further, the door phone sub-unit S includes a microphone 1 ′, a speaker 2 ′, a two-wire / four-wire conversion circuit 3 ′, a microphone amplifier G1 ′, and a speaker amplifier G4 ′.
[0004]
The voice switch 10 'includes a transmission side attenuator 11 for inserting a loss in the signal path on the transmission side, a reception side attenuator 12 for inserting a loss in the signal path on the reception side, and each of the transmission side and the reception side. And an insertion loss amount control unit 13 ′ for controlling the loss amount inserted from the attenuators 11 and 12. The insertion loss amount control unit 13 ′ receives the acoustic echo path H from the output point Rout of the receiving side attenuator 12. _AC Is used to estimate the acoustic feedback gain of the path returning to the input point Tin of the transmitting side attenuator 11 (hereinafter referred to as “acoustic feedback path”), and from the output point Tout of the transmitting side attenuator 11 to the line Echo path H _LIN Is used to estimate the line-side feedback gain of the path returning to the input point Rin of the receiver-side attenuator 12 (hereinafter referred to as “line-side feedback path”), and based on the estimated values of the feedback gains on the acoustic side and the line side The sum of the loss amounts to be inserted into the closed loop (the sum of the insertion loss amount of the transmitting side attenuator 11 and the insertion loss amount of the receiving side attenuator 12) is calculated, and the transmission signal and the reception signal are monitored to make a call. The state is estimated, and the distribution of the insertion loss amounts of the transmitting side attenuator 11 and the receiving side attenuator 12 is determined according to the estimation result and the calculated value of the total insertion loss amount.
[0005]
[Problems to be solved by the invention]
By the way, when the level difference between the ambient noise on the far end side (door phone slave unit S side which is the other party's call terminal) and the ambient noise on the near end side (microphone 1 side) is large, the transmission signal and the reception signal are monitored. Then, the insertion loss amount control unit 13 ′ of the voice switch 10 ′ that estimates the call state always determines that the receiving state is in a situation where the ambient noise level on the far end side is large, and the ambient noise level on the near end side is high. In a situation, the state is always determined as the transmission state, and the call state is fixed to either the reception state or the transmission state regardless of the actual call state (so-called voice switch 10 'fall down) May occur.
[0006]
In contrast, the present inventors estimate a far-end background noise level estimation unit for estimating the far-end background noise level included in the received signal and a near-end background noise level included in the transmitted signal. The near-end side background noise level estimating unit, the reception bias mode setting amplifier for amplifying the reception signal monitored by the insertion loss amount control unit 13 ′, and the transmission monitored by the insertion loss amount control unit 13 ′. Transmitter bias mode setting amplifier that amplifies the signal, and each gain of the receiver bias mode setting amplifier and transmitter bias mode setting amplifier according to the estimated values of the far-end side background noise level and the near-end side background noise level An eccentric mode control unit for adjusting the transmission mode, and if the estimated value of the far-end side background noise level is sufficiently larger than the estimated value of the near-end side background noise level, Set amplifier gain The call processing means is set to the transmission bias mode by increasing the gain of the speech bias mode setting amplifier, and the estimated value of the near-end side background noise level is sufficiently larger than the estimated value of the far-end side background noise level. If so, the gain of the reception bias mode setting amplifier is set to be larger than the gain of the transmission bias mode setting amplifier to set the speech processing means to the reception bias mode, and the far-end side background noise level estimate and the near-end side are set. If the difference between the estimated values of the background noise level is not a sufficiently large value, the gains of the reception bias mode setting amplifier and the transmission bias mode setting amplifier are set to approximately zero to set the speech processing means to the neutral mode, thereby Even if there is a large difference between the ambient noise level at the end-side call terminal and the ambient noise level at the near-end loudspeaker (master unit M ′), the direction of the call is overwhelmed by the higher ambient noise level. Already filed those prevented from (see Japanese Patent Application No. 2001-175521).
[0007]
However, in the above-mentioned loudspeaker apparatus by the present inventors, for example, when the transmission bias mode is set, the gain of the transmission bias mode setting amplifier is increased, so that so-called reception blocking is likely to occur, or when the reception bias mode is set. Since the gain of the reception bias mode setting amplifier is increased, so-called transmission blocking is likely to occur. Here, reception blocking refers to an amplifier for setting a transmission eccentricity mode of an acoustic echo signal generated by acoustic coupling between a speaker and a microphone on the near end when sound is input from the far end while the near end is silent. The power level at the output point of the input signal becomes higher than the power level at the output point of the amplifier for setting the reception bias mode of the original signal (received signal), and the voice switch enters the transmission state. This refers to a phenomenon in which the received sound cannot be heard at the near end. Also, transmission blocking is the setting of the reception eccentricity mode for line echo signals caused by acoustic coupling at the far end or wraparound of signals in the transmission processing means when speech is input from the near end while the far end is silent. When the power level at the output point of the amplifier for the power source becomes higher than the power level at the output point of the amplifier for setting the transmission bias mode of the original signal (transmission signal) and the voice switch is in the receiving state, the near end This is a phenomenon in which voice input from the side cannot be heard at the far end.
[0008]
The present invention has been made in view of the above circumstances, and an object of the present invention is to provide a loudspeaker device that prevents a voice switch from falling over without causing transmission blocking and reception blocking.
[0009]
[Means for Solving the Problems]
In order to achieve the above object, the invention of claim 1 suppresses howling by a microphone that outputs collected sound as a transmission signal, a speaker that rings in response to a reception signal from the other party's call terminal, and the like. Call processing means for enabling a loud voice call, and the call processing means includes an acoustic echo path in which an acoustic echo is generated by acoustic coupling of a microphone and a speaker, and an acoustic coupling or transmission processing means of a signal at a partner telephone terminal. And a voice switch that suppresses howling by reducing a loop gain of a closed loop formed by a line echo path in which a line echo is generated by wraparound, and the voice switch inserts a loss in the signal path on the transmitter side. The loss insertion means, the reception side loss insertion means for inserting loss into the signal path on the receiver side, and the levels of the transmission signal and the reception signal are compared. A loudspeaker communication device comprising an insertion loss amount control means for controlling a loss amount inserted from each loss insertion means on the transmission side and the reception side, wherein the call processing means includes a background on the far end side included in the reception signal The far-end background noise level estimation unit for estimating the noise level, the near-end background noise level estimation unit for estimating the near-end background noise level included in the transmission signal, and the insertion loss amount control means A reception bias mode setting amplifier that amplifies the received signal, a transmission bias mode setting amplifier that amplifies the transmission signal monitored by the insertion loss amount control means, and an acoustic echo from the output point of the reception loss insertion means An acoustic-side feedback gain estimator for estimating the acoustic-side feedback gain of the path that returns to the input point of the transmission-side loss insertion means via the path, and an incoming call from the output point of the transmission-side loss insertion means via the line echo path Side loss insertion means A line-side feedback gain estimator that estimates the line-side feedback gain of the path returning to the point, and according to the estimated values of the far-end background noise level, the near-end background noise level, the acoustic-side feedback gain, and the line-side feedback gain A bias mode control unit that adjusts the gains of the reception bias mode setting amplifier and the transmission bias mode setting amplifier, and the bias mode control unit has a far-end side background noise level estimated value of the near-end side background noise. Than the estimated level Enough If the state where the estimated value of the acoustic feedback gain is smaller than the predetermined value continues for a certain time or longer, the gain of the transmission bias mode setting amplifier is increased more than the gain of the reception bias mode setting amplifier. The call processing means is set to the transmission bias mode, and the estimated value of the near-end background noise level is higher than the estimated value of the far-end background noise level. Enough If the state where the estimated value of the line-side feedback gain is smaller than the predetermined value continues for a certain time or more, the gain of the reception bias mode setting amplifier is increased more than the gain of the transmission bias mode setting amplifier. And set the call processing means to the listening bias mode, and the difference between the far-end background noise level estimate and the near-end background noise level estimate Absolute value of But Enough If the state of a large value does not continue for a certain time or longer, the call processing means is set to the neutral mode by not amplifying the reception signal and the transmission signal by the reception deviation mode setting amplifier and the transmission deviation mode setting amplifier. The feature is set to transmit decent mode, receive decent mode, and neutral mode, considering not only the difference in ambient noise level between the far-end and near-end sides, but also the estimated values of the acoustic feedback gain and line feedback gain. Therefore, it is possible to prevent the voice switch from falling down without causing transmission blocking and reception blocking.
[0010]
According to a second aspect of the present invention, in the first aspect of the invention, the bias mode control unit sets the gains of the reception bias mode setting amplifier and the transmission bias mode setting amplifier as the estimated values of the acoustic side feedback gain and the line side feedback gain. The voice sound pressure level on the far end side or near end side necessary for switching the voice switch to the reception state or the transmission state can be lowered.
[0011]
According to a third aspect of the present invention, in the first aspect of the invention, the deviation mode control unit determines the gains of the reception deviation mode setting amplifier and the transmission deviation mode setting amplifier as the far-end side background noise level and the near-end side background noise. The voice switch can be switched in a balanced manner between the transmission state and the reception state.
[0012]
DETAILED DESCRIPTION OF THE INVENTION
FIG. 1 is a block diagram showing a loudspeaker apparatus according to this embodiment. In this embodiment, a loudspeaker device is configured as the master unit M of the hands-free intercom described in the conventional example, and the basic configuration of this embodiment is the earlier application (Japanese Patent Application) filed by the present inventors. 2001-175521). In addition, in FIG. 1, illustration of the door phone cordless handset S which is an other party call terminal is abbreviate | omitted.
[0013]
In this embodiment, a microphone 1, a speaker 2, a 2-wire to 4-wire conversion circuit 3, a microphone amplifier G1, a line output amplifier G2 for amplifying a transmission signal to a line (two-wire transmission line), and a reception signal from the line A line input amplifier G3, a speaker amplifier G4, a transmission volume adjustment amplifier G5, a reception volume adjustment amplifier G6, first and second echo cancellers 30A and 30B, and a voice switch 10.
[0014]
The first echo canceler 30A has a conventionally known configuration including an adaptive filter 31A and a subtractor 32A, and an acoustic echo path H formed by acoustic coupling between the speaker 2 and the microphone 1. _AC 1 is adaptively identified by the adaptive filter 31A, and an echo component (acoustic echo) estimated from the reference signal (input signal to the speaker amplifier G4) is output from the microphone amplifier G1 by the subtractor 32A (point in FIG. 1). By subtracting from (A signal), the echo component is canceled and eliminated. The second echo canceller 30B also has a conventionally well-known configuration including an adaptive filter 31B and a subtractor 32B, and has reflection and doorphone elements due to impedance mismatch between the 2-wire / four-wire conversion circuit 3 and the transmission path. Line echo path H formed by acoustic coupling between speaker 2 'and microphone 1' in machine S _LIN 1 is adaptively identified by the adaptive filter 31B, and an echo component (line echo) estimated from the reference signal (input signal to the line output amplifier G2, that is, transmission signal) is received by the subtractor 32B (see FIG. 1). By subtracting it from the signal at point C), the echo component is canceled and eliminated.
[0015]
The voice switch 10 includes a transmission side attenuator 11 as a transmission side loss insertion means for inserting a loss into the signal path on the transmission side, and a reception side attenuation as a reception side loss insertion means for inserting a loss into the signal path on the reception side. And an insertion loss amount control unit 13 for controlling the amount of loss inserted from each of the attenuators 11 and 12 on the transmission side and the reception side. The insertion loss amount control unit 13 receives the acoustic echo path H from the output point Rout of the reception side attenuator 12. _AC Is used to estimate the acoustic feedback gain α of the path returning to the input point Tin of the transmission side attenuator 11 (hereinafter referred to as “acoustic feedback path”) and from the output point Tout of the transmission side attenuator 11. Line echo path H _LIN The line-side feedback gain β of the path returning to the input point Rin of the receiving-side attenuator 12 (hereinafter referred to as “line-side feedback path”) is estimated, and the feedback gains α and β on the acoustic side and the line side are estimated. Total loss calculation for calculating the total loss amount to be inserted into the closed loop based on the estimated values α ′ and β ′ (the sum of the insertion loss amount of the transmission side attenuator 11 and the insertion loss amount of the reception side attenuator 12). The communication state is estimated by monitoring the transmission signal and the reception signal, and each of the transmission side attenuator 11 and the reception side attenuator 12 according to the estimation result and the calculated value of the total loss calculation unit 14. And an insertion loss amount distribution processing unit 15 for determining the distribution of the insertion loss amount. The call processing means including the first and second echo cancellers 30A and 30B and the voice switch 10 can be realized by a conventionally known technique using a DSP (Digital Signal Processor).
[0016]
The total loss amount calculation unit 14 estimates the time-average power of the input signal of the transmission side attenuator 11 in a short time using a rectifier / smoothing device, a low-pass filter, and the like. Is used to estimate the time average power of the output signal of the receiving side attenuator 12 in a short time, and the acoustic side feedback path H _AC The minimum value of the estimated value of the time average power of the output signal of the reception side attenuator 12 is obtained at the maximum delay time assumed in FIG. 1, and the estimated value of the time average power of the input signal of the transmission side attenuator 11 is obtained with this minimum value. Is the estimated value α ′ of the acoustic feedback gain α, and the time average power of the input signal of the receiver attenuator 12 in a short time is estimated using a rectifier smoother, a low-pass filter, or the like, Similarly, the time average power of the output signal of the transmission side attenuator 11 in a short time is estimated using a rectifier / smoothing device, a low-pass filter, etc., and the line side feedback path H _LIN , The minimum value of the estimated value of the time average power of the output signal of the transmitting side attenuator 11 is obtained at the maximum delay time assumed in FIG. Is the estimated value β ′ of the line-side feedback gain β. Then, the total loss calculation unit 14 calculates a total loss Lt necessary to obtain a desired gain margin MG from the estimated values α ′ and β ′ of the acoustic feedback gain α and the line feedback gain β. The value Lt is output to the insertion loss amount distribution processing unit 15.
[0017]
The insertion loss amount distribution processing unit 15 monitors the input / output signals of the transmitting side attenuator 11 and the input / output signals of the receiving side attenuator 12, and information such as the magnitude relationship between the power levels of these signals and the presence / absence of an audio signal. The communication state (the reception state, the transmission state, etc.) is determined from the transmission state, and the total loss Lt is distributed to the transmission side attenuator 11 and the reception side attenuator 12 at a rate corresponding to the determined call state. The amount of insertion loss of the attenuators 11 and 12 is adjusted.
[0018]
By the way, as described above, the total loss amount calculation unit 14 according to the present embodiment calculates and adaptively updates the sum of loss amounts to be inserted into the closed loop based on the estimated values α ′ and β ′ of the feedback gains α and β. There are two operation modes: an update mode and a fixed mode for fixing the total loss amount to a predetermined initial value, and the first and second echo cancellers 30A from the start of a call with the other party's call terminal (doorphone slave unit S). , 30B operates in a fixed mode during a period until it sufficiently converges, and operates in an update mode during a period after the first and second echo cancellers 30A, 30B sufficiently converge. That is, in the total loss amount calculation unit 14, the estimated values α ′ and β ′ of the acoustic side feedback gain α and the line side feedback gain β are continuously maintained for a predetermined time (several hundred milliseconds) for a predetermined threshold value ε ( For example, it is considered that the first and second echo cancellers 30A and 30B have sufficiently converged when the values are less than 10 dB to 15 dB smaller than the estimated values α ′ and β ′ at the start of the call, Before, the operation mode is switched to the update mode in which the total loss amount is adaptively updated based on the estimated values α ′ and β ′. Note that the initial value of the total loss amount in the fixed mode is set to a value sufficiently larger than the total loss amount updated as needed in the update mode.
[0019]
Thus, when the first and second echo cancellers 30A and 30B immediately after the start of the call are not sufficiently converged, the total loss amount calculation unit 14 operating in the fixed mode sets the value sufficiently large. Since the initial total loss amount is inserted into the closed loop, it is possible to suppress the generation of unpleasant echoes (acoustic echoes and line echoes) and howling, and realize a stable half-duplex call. In the state where the first and second echo cancellers 30A and 30B have sufficiently converged after the time from the start of the call, the operation mode of the total loss calculation unit 14 is switched from the fixed mode to the update mode and closed loop. Since the total loss amount to be inserted into the value decreases to a value sufficiently lower than the initial value, two-way simultaneous calls can be realized. In addition, by setting the initial value of the total loss amount to an appropriate value, the closed loop loop gain is reduced in order to prevent howling in the state where the first and second echo cancellers 30A and 30B have not converged immediately after the start of the call. There is no restriction of designing the gain of each amplifier so that it does not exceed 1 time, and the gain of the amplifier is obtained so that a desired call volume can be obtained regardless of the shape or structure of the housing (not shown) of the main unit M. Can be designed.
[0020]
Next, a specific operation of the total loss amount calculation unit 14 in the update mode will be described with reference to the flowchart of FIG.
[0021]
The total loss amount calculation unit 14 executes an estimation process of the acoustic side feedback gain α and the line side feedback gain β at a predetermined sampling period from the time when the fixed mode is changed to the update mode (t = t1), and the estimated value α '(n), β' (n) is calculated (step 1), and the gain margin of the closed loop MG [is calculated from the product of these two estimated values α '(n), β' (n) and the gain margin MG. The desired total loss amount Lr (n) required for maintaining the value [dB] is calculated by the following equation (step 2).
[0022]
Lr (n) = 20 log | α ′ (n) · β ′ (n) | + MG [dB]
Note that α ′ (n), β ′ (n), and Lr (n) indicate an estimated value of feedback gain and a desired total loss amount calculated by sampling n times from the update mode transition point, respectively. Further, the total loss amount calculation unit 14 calculates the n-th total loss amount desired value Lr (n) calculated from the above formula and the previous (n−1) th total loss amount Lt (n−1), that is, the previous process. When the desired total loss amount Lr (n) calculated this time is larger than the total loss amount determined and actually inserted, a slight increase Δi [dB in the previous total loss amount Lt (n−1). ] Is defined as the total loss amount Lt (n) = Lt (n−1) + Δi (steps 3 and 4), and the total loss calculated this time with respect to the previous total loss amount Lt (n−1). When the loss desired value Lr (n) is small, the current total loss Lt (n) = Lt (n) is obtained by subtracting a slight decrease Δd [dB] from the previous total loss Lt (n−1). −1) −Δd (steps 5 and 6).
[0023]
In this way, by suppressing the increase / decrease in the total loss amount by the total loss amount calculation unit 14 to a small value of Δi or Δd, the first and the first and the first steps are performed immediately after the start of a call with the other party's call terminal (doorphone slave unit S). Since the two echo cancellers 30A and 30B actively update the coefficients toward convergence, it is possible to eliminate a sense of incongruity even in a state in which the acoustic side feedback gain α and the line side feedback gain β change drastically. .
[0024]
On the other hand, the gist of the present embodiment is that the far-end side background noise level estimation unit 20 that estimates the background noise level on the far-end side (the other party's call terminal) included in the received signal and the near-end included in the transmitted signal A near-end-side background noise level estimation unit 21 that estimates the background noise level on the side (microphone 1 side), a reception bias mode setting amplifier GR that amplifies the reception signal monitored by the insertion loss amount control unit 13, and an insertion A transmission bias mode setting amplifier GT that amplifies the transmission signal monitored by the loss amount control unit 13, and the acoustic echo path H from the output point Rout of the reception side attenuator 12. _AC An acoustic-side feedback gain estimation unit 22 for estimating the acoustic-side feedback gain α of the path returning to the input point Tin of the transmitting-side attenuator 11, and the line echo path H from the output point Tout of the transmitting-side attenuator 11. _LIN A line-side feedback gain estimator 23 for estimating the line-side feedback gain β of the path that returns to the input point Rin of the receiver-side attenuator 12 via the far-end side background noise level, the near-end side background noise level, and the acoustic side A bias mode control unit 24 that adjusts the gains of the reception bias mode setting amplifier GR and the transmission bias mode setting amplifier GT according to the estimated values of the feedback gain and the line side feedback gain, respectively. It is in the point provided for the telephone call processing means including the echo cancellers 30A and 30B and the voice switch 10. The far-end side background noise level estimation unit 20, the near-end side background noise level estimation unit 21, the acoustic-side feedback gain estimation unit 22, the line-side feedback gain estimation unit 23, and the bias mode control unit 24 are the first and second modes. This is realized using a DSP together with the echo cancellers 30A and 30B and the voice switch 10.
[0025]
Each of the far-end side background noise level estimation unit 20 and the near-end side background noise level estimation unit 21 is realized by an integration circuit or a digital filter having a characteristic that the rise is gradual and the fall is steep. The background noise level estimator 20 estimates the background noise level constantly present in the received signal output from the received sound volume adjustment amplifier G6, and the near-end background noise level estimator 21 transmits the transmitted sound volume. A background noise level that is constantly present in the transmission signal output from the adjustment amplifier G5 is estimated. The estimation process of each feedback gain in the acoustic side feedback gain estimation unit 22 and the line side feedback gain estimation unit 23 is the same as the estimation process by the total loss calculation unit 14.
[0026]
The bias mode control unit 24 is such that the estimated value PFn of the far-end side background noise level is sufficiently larger than the estimated value PNn of the near-end side background noise level (PFn >> PNn) and the estimated value of the acoustic feedback gain α. If the state where | α ′ | is smaller than the predetermined value α0 (| α ′ | <α0) continues for a certain time T1 [seconds] or longer, the gain of the transmission bias mode setting amplifier GT is set to G1 [dB]. (G1> 0), setting the gain of the reception bias mode setting amplifier GR to 0 [dB] sets the voice switch 10 to the transmission bias mode, and the near-end side background noise level estimation value PNn is the far-end side. The estimated value | β ′ | of the line side feedback gain β is a value sufficiently larger than the estimated value PFn of the background noise level (PNn >> PFn) and smaller than the predetermined value β0 (| β ′ | <β0). If the state continues for a certain period of time T2 [seconds] or longer, the reception bias By setting the gain of the mode setting amplifier GR to G2 [dB] (G2> 0) and the gain of the transmission bias mode setting amplifier GT to 0 [dB], the voice switch 10 is set to the reception bias mode. If the state where the difference between the estimated value PFn of the end side background noise level and the estimated value PNn of the near end side background noise level is sufficiently large does not continue for a certain time T1 or T2 or more, the receiver GR mode setting amplifier GR and The voice switch 10 is set to the neutral mode by not amplifying the reception signal and the transmission signal by the transmission bias mode setting amplifier GT, that is, by making each gain substantially zero.
[0027]
That is, when the difference between the ambient noise level on the far end side and the ambient noise level on the near end side is large, the insertion loss amount control unit 13 that monitors the transmission signal and the received signal to estimate the call state, for example, In the situation where the ambient noise level on the side is large, it is always determined as the reception state, and in the situation where the near-end ambient noise level is large, it is always determined as the transmission state, regardless of the actual call state. There is a case in which a call state is fixed to one of the transmission states (so-called voice switch 10 is tilted down). In contrast, the inventors of the present invention compare the estimated value PFn of the far-end side background noise level with the estimated value PNn of the near-end side background noise level, and the biased mode control unit 24 compares the estimated value PNn of the far-end side background noise level. If PFn is sufficiently larger, the insertion loss amount control unit 13 transmits the transmission signal monitored by the insertion loss amount control unit 13 by amplifying the transmission bias mode setting amplifier GT by a gain G1 [dB]. If the estimated value PNn of the near-end side background noise level is sufficiently larger, on the contrary, the received signal monitored by the insertion loss amount control unit 13 is received. By amplifying only the gain G2 [dB] by the mode setting amplifier GR, the insertion loss amount control unit 13 is set to a state in which it is easy to determine the reception state (reception deviation mode), thereby preventing the voice switch 10 from falling down. A voice communication device has already proposed. However, the deviation mode control unit 24 switches between the transmission deviation mode, the reception deviation mode, and the neutral mode based only on the comparison result between the far-end side background noise level estimation value PFn and the near-end side background noise level estimation value PNn. For example, when the acoustic feedback gain α is a large value to some extent, a phenomenon in which the receiving state is supposed to be in the original state (receiving blocking) or the line side feedback gain β is somewhat large. When the value is a value, there is a possibility that a phenomenon (transmission blocking) that does not become a transmission state although it should originally be in a transmission state may occur.
[0028]
Therefore, in the present invention, the bias mode control unit 24 not only calculates the difference between the estimated value PFn of the far-end side background noise level and the estimated value PNn of the near-end side background noise level, but also the acoustic-side feedback gain α and the line-side feedback gain. Considering each estimated value | α ′ | and | β ′ | of β, the transmission bias mode, the reception bias mode, and the neutral mode are set. That is, even when the situation where the estimated value PFn of the far-end side background noise level is sufficiently larger than the estimated value PNn of the near-end side background noise level continues for a certain time T1 or longer, the estimated value | α ′ of the acoustic-side feedback gain α When | exceeds a predetermined value α0, the switch is not shifted to the transmission bias mode, so that the voice switch 10 can be prevented from falling down without causing reception blocking. Even when the estimated value PNn of the near-end background noise level is sufficiently larger than the estimated value PFn of the far-end background noise level continues for a certain time T2 or longer, the estimated value | β ′ of the line-side feedback gain β When | exceeds a predetermined value β0, the shift to the reception bias mode is not performed, so that it is possible to prevent the voice switch 10 from falling down without causing transmission blocking.
[0029]
Here, in order to prevent reception blocking and transmission blocking, | α ′ | + G1 <M1, | β ′ | + G2 <M2 (rewriting this, | α ′ | <M1-G1 = α0, | β The relationship of '| <M2-G2 = β0) must be established. That is, the predetermined values α0 and β0 are determined according to reception blocking, margins M1 and M2 for transmission blocking, gain G1 of the transmission bias mode setting amplifier GT, and gain G2 of the reception bias mode setting amplifier GR. It should be noted that the gain G1 of the transmission bias mode setting amplifier GT and the gain G2 of the reception bias mode setting amplifier GR are set to a near-end background noise level assumed in the usage environment, respectively. It is determined on the basis of the required specification as a loudspeaker device for switching the voice switch 10 to the transmission state or the reception state when the sound pressure level is spoken from the side and the far end side.
[0030]
By the way, when the deviation mode control unit 24 switches from the neutral mode to the transmission deviation mode or the reception deviation mode, the gains G1 and G2 of the transmission deviation mode setting amplifier GT and the reception deviation mode setting amplifier GR are fed back to the acoustic side. The gain α and the line-side feedback gain β may be increased or decreased in stages according to the estimated values | α ′ | and | β ′ |. For example, if the gain G1 is increased / decreased in three stages, the deflection mode control unit 24 determines that the estimated value | α ′ | of the acoustic feedback gain α is | α ′ | ≧ α0. ₁ When GT = 0, α0 ₂ ≦ | α ′ | <α0 ₁ When GT = G1 ″, α0 _Three ≦ | α ′ | <α0 ₂ When GT = G1 ′, | α ′ | <α0 _Three In this case, the gain G1 of the transmission bias mode setting amplifier GT is increased or decreased so that GT = G1 (where α0 _Three <Α0 ₂ <Α0 ₁ 0 <G1 ″ <G1 ′ <G1) Similarly, if the gain G2 is increased or decreased in three stages, the bias mode control unit 24 determines that the estimated value | β ′ | of the line-side feedback gain β is | β ′ | ≧ β0 ₁ When GR = 0, β0 ₂ ≦ | β ′ | <β0 ₁ When GR = G2 ″, β0 _Three ≦ | β ′ | <β0 ₂ When GR = G2 ′, | β ′ | <β0 _Three The gain G2 of the reception bias mode setting amplifier GR is increased or decreased so that GR = G2 (note that β0 _Three <Β0 ₂ <Β0 ₁ 0 <G2 ”<G2 ′ <G2).
[0031]
For example, when the gain G1 of the transmission bias mode setting amplifier GT is set to a large value, the predetermined value α0 is set to a small value, and the estimated value | α ′ | When the initial value is large or when the convergence speed of the first acoustic echo canceller 30A is slow, the time required for the estimated value | α ′ | to fall below the predetermined value α0 becomes long. In some cases, | α ′ | does not reach a state where it falls below a predetermined value α0. Even in such a case, as described above, when the deviation mode control unit 24 switches from the neutral mode to the transmission deviation mode, the gain G1 of the transmission deviation mode setting amplifier GT is set to the estimated value of the acoustic feedback gain α. If it is increased or decreased step by step in accordance with α ′ |, the voice pressure level on the near end necessary for switching the voice switch 10 to the transmission state can be lowered, and the voice switch 10 is switched to the transmission state. It becomes easy to change. Similarly, when the gain G2 of the reception bias mode setting amplifier GR is set to a large value, the predetermined value β0 is set to a small value, and the estimated value | β ′ | When the initial value is large or when the convergence speed of the second acoustic echo canceller 30B is slow, the time required for the estimated value | β ′ | to fall below the predetermined value β0 becomes long. In some cases, | β ′ | does not reach a state where it falls below a predetermined value β0. Even in such a case, when the bias mode control unit 24 switches from the neutral mode to the reception bias mode as described above, the gain G2 of the reception bias mode setting amplifier GR is set to the estimated value | β ′ of the line-side feedback gain β. If it is increased or decreased step by step in accordance with |, the far-end utterance sound pressure level necessary for switching the voice switch 10 to the reception state can be lowered, and the voice switch 10 can easily be switched to the reception state.
[0032]
Further, when the bias mode controller 24 switches from the neutral mode to the transmission bias mode or the reception bias mode, the gains G1 and G2 of the transmission bias mode setting amplifier GT and the reception bias mode setting amplifier GR are set to the far end side. You may make it increase / decrease in steps according to the difference between the estimated value PFn of the background noise level and the estimated value PNn of the near-end side background noise level. For example, if the gain G1 is increased / decreased in three stages, the bias mode controller 24 determines that GT = 0 when the difference X (= PFn−PNn) is X ≦ X1, and GT = when X1 <X ≦ X2. The gain G1 of the transmission bias mode setting amplifier GT is increased or decreased so that GT = G1 ′ when G1 ″, X2 <X ≦ X3, and GT = G1 when X> X3 (however, X1 <X2 < X3, 0 <G1 "<G1 '<G1). Similarly, if the gain G2 is increased or decreased in three stages, the bias mode control unit 24 determines that GR = 0 when the difference Y (= PNn−PFn) is Y ≦ Y1, and GR = when Y1 <Y ≦ Y2. The gain G2 of the reception bias mode setting amplifier GR is increased or decreased so that GR = G2 ′ when G2 ″, Y2 <Y ≦ Y3, and GR = G2 when Y> Y3 (however, Y1 <Y2 <Y3 0 <G2 ”<G2 ′ <G2).
[0033]
As described above, when the deviation mode control unit 24 switches from the neutral mode to the transmission deviation mode or the reception deviation mode, the gains G1 and G2 of the transmission deviation mode setting amplifier GT and the reception deviation mode setting amplifier GR are set to be far. By increasing or decreasing in steps according to the difference between the estimated value PFn of the end side background noise level and the estimated value PNn of the near end side background noise level, for example, the estimated value PFn of the far end side background noise level and the near end When the difference from the estimated value PNn of the side background noise level is large and the set values of the gains G1 and G2 are large, the voice switch 10 is switched between the transmission state and the reception according to the background noise level difference between the near end side and the far end side. It is possible to switch to the state in a balanced manner.
[0034]
【The invention's effect】
According to the first aspect of the present invention, there is provided a microphone that outputs the collected sound as a transmission signal, a speaker that rings in response to a reception signal from the other party's telephone terminal, and a call that enables a voice call while suppressing howling. A speech processing unit including an acoustic echo path in which an acoustic echo is generated due to acoustic coupling between a microphone and a speaker, and a line echo in which a line echo is generated due to acoustic coupling or signal wraparound in a transmission processing unit on the other party's telephone terminal. A voice switch that reduces a loop gain of the closed loop formed by the path and suppresses howling, and the voice switch includes transmission-side loss insertion means for inserting a loss into the signal path on the transmission side, Receiver side loss insertion means for inserting loss in the signal path, and each loss on the transmitter side and receiver side by comparing the levels of the transmitted signal and the received signal A speech communication apparatus comprising an insertion loss amount control means for controlling a loss amount inserted from an input means, wherein the call processing means estimates a background noise level on the far end side included in the received signal. Noise level estimator, near-end background noise level estimator that estimates the near-end background noise level included in the transmitted signal, and reception bias mode that amplifies the received signal monitored by the insertion loss control means Setting amplifier, transmission bias mode setting amplifier that amplifies the transmission signal monitored by the insertion loss amount control means, and transmission side loss insertion from the output point of the reception side loss insertion means via the acoustic echo path An acoustic-side feedback gain estimator for estimating the acoustic-side feedback gain of the path returning to the input point of the means, and feedback from the output point of the transmitting-side loss insertion means to the input point of the receiving-side loss insertion means via the line echo path Line side of the route A line-side feedback gain estimator for estimating a return gain, an amplifier for setting a reception bias mode according to estimated values of a far-end side background noise level, a near-end side background noise level, an acoustic-side feedback gain, and a line-side feedback gain; A bias mode control unit that adjusts each gain of the transmission bias mode setting amplifier, and the bias mode control unit has an estimated value of the far-end side background noise level higher than an estimated value of the near-end side background noise level. Enough If the state where the estimated value of the acoustic feedback gain is smaller than the predetermined value continues for a certain time or longer, the gain of the transmission bias mode setting amplifier is increased more than the gain of the reception bias mode setting amplifier. The call processing means is set to the transmission bias mode, and the estimated value of the near-end background noise level is higher than the estimated value of the far-end background noise level. Enough If the state where the estimated value of the line-side feedback gain is smaller than the predetermined value continues for a certain time or more, the gain of the reception bias mode setting amplifier is increased more than the gain of the transmission bias mode setting amplifier. And set the call processing means to the listening bias mode, and the difference between the far-end background noise level estimate and the near-end background noise level estimate Absolute value of But Enough If the state of a large value does not continue for a certain time or longer, the call processing means is set to the neutral mode by not amplifying the reception signal and the transmission signal with the reception deviation mode setting amplifier and the transmission deviation mode setting amplifier. In consideration of not only the difference in the level of ambient noise between the far-end and near-end sides, but also the estimated values of the acoustic feedback gain and line-side feedback gain, it is set to the transmission eccentric mode, reception eccentric mode, and neutral mode. Therefore, there is an effect that the voice switch can be prevented from falling over without causing transmission blocking and reception blocking.
[0035]
According to a second aspect of the present invention, in the first aspect of the invention, the bias mode control unit sets the gains of the reception bias mode setting amplifier and the transmission bias mode setting amplifier as the estimated values of the acoustic side feedback gain and the line side feedback gain. Therefore, the sound pressure level at the far end or near end required for switching the voice switch to the receiving state or the transmitting state can be lowered.
[0036]
According to a third aspect of the present invention, in the first aspect of the invention, the deviation mode control unit determines the gains of the reception deviation mode setting amplifier and the transmission deviation mode setting amplifier as the far-end side background noise level and the near-end side background noise. Since the level is increased or decreased stepwise according to the estimated value of the level, there is an effect that the voice switch can be switched in a balanced manner between the transmission state and the reception state.
[Brief description of the drawings]
FIG. 1 is a block diagram illustrating an embodiment.
FIG. 2 is a flowchart for explaining the operation of the above.
FIG. 3 is a block diagram showing a conventional example.
[Explanation of symbols]
1 Microphone
2 Speaker
10 Voice switch
13 Insertion loss control unit
20 Far-end side background noise level estimation unit
21 Near-end background noise level estimation unit
22 Acoustic side feedback gain estimator
23 Line-side feedback gain estimator
24 Unbalance mode controller

Claims

集音した音声を送話信号として出力するマイクロホンと、相手側の通話端末からの受話信号に応じて鳴動するスピーカと、ハウリングを抑制して拡声通話を可能とする通話処理手段とを備え、通話処理手段は、マイクロホンとスピーカの音響結合によって音響エコーが生じる音響エコー経路、並びに相手側の通話端末における音響結合又は伝送処理手段における信号の回り込みによって回線エコーが生じる回線エコー経路により形成される閉ループの一巡利得を低減してハウリングを抑制する音声スイッチとを有し、音声スイッチは、送話側の信号経路に損失を挿入する送話側損失挿入手段と、受話側の信号経路に損失を挿入する受話側損失挿入手段と、送話信号と受話信号のレベルを比較して送話側及び受話側の各損失挿入手段から挿入する損失量を制御する挿入損失量制御手段とを具備する拡声通話装置であって、通話処理手段は、受話信号に含まれる遠端側の背景雑音レベルを推定する遠端側背景雑音レベル推定部と、送話信号に含まれる近端側の背景雑音レベルを推定する近端側背景雑音レベル推定部と、挿入損失量制御手段で監視している受話信号を増幅する受話偏重モード設定用増幅器と、挿入損失量制御手段で監視している送話信号を増幅する送話偏重モード設定用増幅器と、受話側損失挿入手段の出力点から音響エコー経路を介して送話側損失挿入手段の入力点へ帰還する経路の音響側帰還利得を推定する音響側帰還利得推定部と、送話側損失挿入手段の出力点から回線エコー経路を介して受話側損失挿入手段の入力点へ帰還する経路の回線側帰還利得を推定する回線側帰還利得推定部と、遠端側背景雑音レベル、近端側背景雑音レベル、音響側帰還利得並びに回線側帰還利得の各推定値に応じて受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を調整する偏重モード制御部とを備え、偏重モード制御部は、遠端側背景雑音レベルの推定値が近端側背景雑音レベルの推定値よりも充分に大きい値であり且つ音響側帰還利得の推定値が所定値よりも小さい値となる状態が一定時間以上継続すれば送話偏重モード設定用増幅器の利得を受話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を送話偏重モードに設定し、近端側背景雑音レベルの推定値が遠端側背景雑音レベルの推定値よりも充分に大きい値であり且つ回線側帰還利得の推定値が所定値よりも小さい値となる状態が一定時間以上継続すれば受話偏重モード設定用増幅器の利得を送話偏重モード設定用増幅器の利得よりも増大させて通話処理手段を受話偏重モードに設定し、遠端側背景雑音レベルの推定値と近端側背景雑音レベルの推定値の差の絶対値が充分に大きい値となる状態が一定時間以上継続しなければ受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器で受話信号及び送話信号を増幅しないことにより通話処理手段を中立モードに設定することを特徴とする拡声通話装置。A microphone that outputs the collected sound as a transmission signal, a speaker that rings in response to a reception signal from the other party's call terminal, and a call processing means that suppresses howling and enables a voice call. The processing means includes a closed loop formed by an acoustic echo path in which an acoustic echo is generated by acoustic coupling of a microphone and a speaker, and a line echo path in which a line echo is generated by an acoustic coupling in a partner telephone terminal or a signal wraparound in a transmission processing means. A voice switch that reduces the loop gain and suppresses howling, and the voice switch inserts a loss in the signal path on the receiving side, and a loss insertion means for inserting the loss in the signal path on the transmitting side. The loss insertion means on the receiving side and the level of the transmitted signal and the received signal are compared and inserted from each loss insertion means on the transmitting side and the receiving side. A speech communication apparatus comprising an insertion loss amount control means for controlling a loss amount, wherein the call processing means includes a far end side background noise level estimation unit for estimating a far end side background noise level included in a received signal; A near-end side background noise level estimation unit for estimating the near-end side background noise level included in the transmission signal, a reception bias mode setting amplifier for amplifying the reception signal monitored by the insertion loss amount control means, A transmission bias mode setting amplifier that amplifies the transmission signal monitored by the insertion loss amount control means, and the input point of the transmission side loss insertion means via the acoustic echo path from the output point of the reception side loss insertion means An acoustic-side feedback gain estimator that estimates the acoustic-side feedback gain of the feedback path, and the line side of the path that returns from the output point of the transmitting-side loss insertion means to the input point of the receiving-side loss insertion means via the line echo path Estimate feedback gain Line-side feedback gain estimator, far-end-side background noise level, near-end-side background noise level, acoustic-side feedback gain, and line-side feedback gain estimation amplifier and transmission-biasing mode setting A bias mode control unit that adjusts each gain of the amplifier for use, and the bias mode control unit has a far-end side background noise level estimate that is sufficiently larger than a near-end side background noise level estimate and If the state in which the estimated value of the acoustic feedback gain is smaller than the predetermined value continues for a certain time or longer, the gain of the transmission bias mode setting amplifier is increased more than the gain of the reception bias mode setting amplifier, and the speech processing means Is set to the transmission bias mode, the estimated value of the near-end background noise level is sufficiently larger than the estimated value of the far-end background noise level, and the estimated value of the line-side feedback gain is smaller than the predetermined value. Value If the state continues for a certain time or longer, the gain of the reception bias mode setting amplifier is set to be larger than the gain of the transmission bias mode setting amplifier to set the speech processing means to the reception bias mode, and the far-end background noise level is estimated. If the state in which the absolute value of the difference between the value and the estimated value of the near-end side background noise level is sufficiently large does not continue for a certain period of time, the received signal and the reception bias mode setting amplifier and the transmission bias mode setting amplifier A loudspeaker apparatus characterized in that the call processing means is set to a neutral mode by not amplifying the transmission signal.

偏重モード制御部は、受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を音響側帰還利得並びに回線側帰還利得の推定値に応じて段階的に増減させることを特徴とする請求項１記載の拡声通話装置。 The deviation mode control unit increases or decreases each gain of the reception deviation mode setting amplifier and the transmission deviation mode setting amplifier stepwise in accordance with the estimated values of the acoustic side feedback gain and the line side feedback gain. Item 2. The voice communication device according to Item 1.

偏重モード制御部は、受話偏重モード設定用増幅器並びに送話偏重モード設定用増幅器の各利得を遠端側背景雑音レベル並びに近端側背景雑音レベルの推定値に応じて段階的に増減させることを特徴とする請求項１記載の拡声通話装置。 The bias mode control unit increases or decreases the gains of the reception bias mode setting amplifier and the transmission bias mode setting amplifier in a stepwise manner in accordance with the estimated values of the far-end side background noise level and the near-end side background noise level. The loudspeaker device according to claim 1, wherein: