JP2007124162A - Loudspeaker call device - Google Patents

Loudspeaker call device Download PDF

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JP2007124162A
JP2007124162A JP2005312003A JP2005312003A JP2007124162A JP 2007124162 A JP2007124162 A JP 2007124162A JP 2005312003 A JP2005312003 A JP 2005312003A JP 2005312003 A JP2005312003 A JP 2005312003A JP 2007124162 A JP2007124162 A JP 2007124162A
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loss
echo
signal
path
reception
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恵一 ▲吉▼田
Keiichi Yoshida
Minoru Fukushima
実 福島
Hiroaki Takeyama
博昭 竹山
Hiroshi Kyomen
公士 京面
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Panasonic Electric Works Co Ltd
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Matsushita Electric Works Ltd
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Abstract

<P>PROBLEM TO BE SOLVED: To provide a loudspeaker call device capable of preventing the occurrence of howling, even if the received speech volume is increased. <P>SOLUTION: The loudspeaker call device comprises a microphone 1; a speaker 2; a 2-4 wire conversion circuit 3; a received signal amplifier VG; a voice switch 10; and first and second echo cancellers 30A, 30B. When a talker manually increases the amplification factor of the received signal amplifier VG to increase the received speech volume, the total loss amount calculation section 14 of the voice switch 10 increases the total loss by an amount nearly equal to the increment of the amplification factor of the received signal amplifier VG. As a result, since the increase in the loop gain of the closed loop formed by the acoustic side feedback path H<SB>AC</SB>and the line side feedback path H<SB>LIN</SB>is suppressed, even if the amplification factor of the received signal amplifier VG is increased, the occurrence of howling can be prevented. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

本発明は、マイクロホン並びにスピーカを具備して拡声通話を行うインターホン等の拡声通話装置に関するものである。   The present invention relates to a loudspeaker communication apparatus such as an interphone that includes a microphone and a speaker and performs a loudspeaker call.

この種の拡声通話装置では、マイクロホンとスピーカの音響結合により形成される音響側の帰還経路や、相手側の通話端末との間で形成される回線側の帰還経路によって不快なエコー(音響エコーあるいは回線エコー)が聞こえてしまう場合があり、あるいは、上記帰還経路などにより任意の周波数成分における一巡利得が1倍を超えるような閉ループが通話系に形成されると当該周波数にてハウリングが生じてしまう場合があるので、上述のような不快なエコー及びハウリングの発生を防止するためにエコーキャンセラ並びに音声スイッチを備えている(例えば、特許文献1参照)。   In this type of loudspeaker, an uncomfortable echo (acoustic echo or acoustic echo) is caused by an acoustic return path formed by acoustic coupling of a microphone and a speaker, or a line-side return path formed between the other party's call terminal. Line echo) may be heard, or howling occurs at the frequency when a closed loop in which the loop gain in an arbitrary frequency component exceeds 1 is formed in the communication system by the feedback path or the like. In some cases, an echo canceller and a voice switch are provided in order to prevent the generation of unpleasant echoes and howling as described above (see, for example, Patent Document 1).

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部と、受話側の信号経路に損失を挿入する受話側損失挿入部と、送話側及び受話側の各損失挿入部から挿入する損失量を制御する挿入損失量制御部とを具備する。挿入損失量制御部は、受話側損失挿入部の出力点から音響エコー経路を介して送話側損失挿入部の入力点へ帰還する経路(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、送話側損失挿入部の出力点から回線エコー経路を介して受話側損失挿入部の入力点へ帰還する経路(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(送話側損失挿入部の挿入損失量と受話側損失挿入部の挿入損失量の和)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損失挿入部及び受話側損失挿入部の各挿入損失量の配分を決定している。また、エコーキャンセラは、帰還経路のインパルス応答を適応的に同定して帰還経路への入力信号から帰還経路の擬似エコー成分を推定する適応フィルタと、適応フィルタで推定された擬似エコー成分を帰還経路からの出力信号より減算する減算器とで構成されるものである。
特開2002−359580号公報
The voice switch includes a transmission side loss insertion unit that inserts loss into the signal path on the transmission side, a reception side loss insertion unit that inserts loss into the signal path on the reception side, and each loss insertion on the transmission side and reception side. And an insertion loss amount control unit for controlling a loss amount inserted from the unit. The insertion loss amount control unit is configured to return the acoustic side feedback (hereinafter referred to as “acoustic side feedback path”) that returns from the output point of the receiving side loss insertion unit to the input point of the transmission side loss insertion unit via the acoustic echo path. Line-side feedback gain of the path (hereinafter referred to as “line-side feedback path”) for estimating the gain and returning from the output point of the transmission-side loss insertion section to the input point of the reception-side loss insertion section via the line echo path And the total loss amount to be inserted into the closed loop based on the estimated values of the feedback gains on the acoustic side and the line side (the insertion loss amount of the transmission side loss insertion unit and the insertion loss amount of the reception side loss insertion unit) Sum)), and the call state is estimated by monitoring the transmission signal and the reception signal, and the transmission side loss insertion unit and the reception side loss insertion unit according to the estimation result and the calculated value of the total insertion loss amount The distribution of each insertion loss amount is determined. The echo canceller adaptively identifies the impulse response of the feedback path and estimates the pseudo echo component of the feedback path from the input signal to the feedback path, and the pseudo echo component estimated by the adaptive filter as the feedback path. And a subtracter that subtracts from the output signal from the.
JP 2002-359580 A

ところで、上述のような拡声通話装置において、高齢者への対応としてスピーカから送出される受話音声の音量(以下、受話音量という。)を手動で大きくする機能を持たせた場合、受話音量が増大することで音響側帰還利得が増えてしまうから、音響エコー経路並びに回線エコー経路により形成される閉ループの一巡利得も増加してハウリングが生じやすくなってしまう。   By the way, in the above-mentioned loudspeaker device, when the function of manually increasing the volume of the received voice (hereinafter referred to as the received volume) sent from the speaker as a response to the elderly, the received volume is increased. As a result, the feedback gain on the acoustic side increases, so that the round-trip gain of the closed loop formed by the acoustic echo path and the line echo path also increases, and howling easily occurs.

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、受話音量を増大したときでもハウリングの発生が防止できる拡声通話装置を提供することにある。   The present invention has been made in view of the above circumstances, and an object of the present invention is to provide a loudspeaker apparatus that can prevent howling even when the received sound volume is increased.

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、マイクロホンとスピーカの音響結合によって生じる音響エコーを消去する第1のエコーキャンセラと、相手側の通話端末における音響結合や相手側の通話端末との間の回線における信号の回り込みによって生じる回線エコーを消去する第2のエコーキャンセラと、第1及び第2のエコーキャンセラの間に設けられ、音響エコー経路並びに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、音声スイッチと第1のエコーキャンセラの間に設けられ、スピーカへ出力される受話信号を増幅し且つ増幅度が可変である受話信号増幅手段とを備え、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し、挿入損失量制御手段は、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とともに、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を算出する総損失量算出部と、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の算出値に応じて送話側損失挿入手段及び受話側挿入損失手段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とからなり、総損失量算出部は、受話信号増幅手段の増幅度が増大した場合に当該増幅度の増大分に略等しい量だけ総損失量を増やすことを特徴とする。   In order to achieve the above object, the invention of claim 1 is a microphone that outputs the collected sound as a transmission signal, a speaker that rings in response to a reception signal from a call terminal on the other side, a microphone and a speaker. A first echo canceler that eliminates acoustic echo caused by acoustic coupling, and a second echo that eliminates line echo caused by signal coupling in the line between the acoustic coupling at the other party's call terminal and the other party's call terminal A voice switch that is provided between the canceller and the first and second echo cancellers, reduces a round-trip gain of a closed loop formed by the acoustic echo path and the line echo path, and suppresses howling; a voice switch; A reception signal that is provided between the echo cancellers, amplifies the reception signal output to the speaker, and has a variable amplification factor. The voice switch includes a transmission side loss insertion unit that inserts a loss into the signal path on the transmission side, a reception side loss insertion unit that inserts a loss into the signal path on the reception side, a transmission side, and Insertion loss amount control means for controlling the loss amount to be inserted from each loss insertion means on the receiver side, the insertion loss amount control means from the output point of the reception side loss insertion means via the acoustic echo path to the transmission side Estimate the acoustic feedback gain of the path returning to the input point of the loss insertion means, and return the line from the output point of the transmission loss insertion means to the input point of the reception loss insertion means via the line echo path A total loss amount calculation unit that estimates the side feedback gain and calculates the total amount of loss to be inserted into the closed loop based on the estimated values of the feedback gains on the acoustic side and the line side, and monitors the transmission signal and the reception signal To estimate the call state and And an insertion loss amount distribution processing unit that determines the distribution of each insertion loss amount of the transmission side loss insertion means and the reception side insertion loss means according to the calculated value of the total loss amount calculation unit, the total loss amount calculation unit is When the amplification level of the reception signal amplification means is increased, the total loss amount is increased by an amount substantially equal to the increase amount of the amplification level.

請求項2の発明は、請求項1の発明において、第2のエコーキャンセラは、回線エコー経路のインパルス応答を適応的に同定して当該回線エコー経路への入力信号から擬似エコー成分を推定する適応フィルタと、適応フィルタで推定された擬似エコー成分を回線エコー経路からの出力信号より減算する減算器とを具備し、適応フィルタは、受話信号増幅手段の増幅度が増大した後も増大前の擬似エコー成分の推定結果を用いることを特徴とする。   According to a second aspect of the present invention, in the first aspect of the invention, the second echo canceller adaptively identifies an impulse response of a line echo path and estimates a pseudo echo component from an input signal to the line echo path. And a subtractor for subtracting the pseudo echo component estimated by the adaptive filter from the output signal from the line echo path, and the adaptive filter is a pseudo signal before the increase after the amplification degree of the received signal amplifying means is increased. The estimation result of the echo component is used.

請求項3の発明は、請求項1又は2の発明において、受話信号増幅手段は、所定の時定数に従って緩やかに増幅度を変化させることを特徴とする。   The invention of claim 3 is characterized in that, in the invention of claim 1 or 2, the received signal amplification means gently changes the amplification degree according to a predetermined time constant.

請求項1の発明によれば、受話信号増幅手段の増幅度を増大して受話音量を大きくした場合においても、音声スイッチの総損失量算出部が、当該増幅度の増大分に略等しい量だけ総損失量を増やすため、音響エコー経路並びに回線エコー経路により形成される閉ループの一巡利得の増加が抑制されてハウリングの発生を防止することができるという効果がある。   According to the first aspect of the present invention, even when the amplification level of the reception signal amplification means is increased to increase the reception volume, the total loss amount calculation unit of the voice switch is an amount substantially equal to the increase of the amplification level. In order to increase the total loss amount, an increase in the closed loop loop gain formed by the acoustic echo path and the line echo path is suppressed, and it is possible to prevent the occurrence of howling.

請求項2の発明によれば、請求項1と同様の効果を奏する。   According to invention of Claim 2, there exists an effect similar to Claim 1.

請求項3の発明によれば、総損失量算出部における音響側帰還利得の推定処理に与える負荷を軽減してハウリングの発生をさらに確実に防止できるという効果がある。   According to the third aspect of the present invention, there is an effect that howling can be more reliably prevented by reducing the load applied to the acoustic side feedback gain estimation process in the total loss amount calculation unit.

本実施形態は、図1に示すようにマイクロホン1、スピーカ2、2線−4線変換回路3、受話信号増幅器VG、音声スイッチ10並びに第1及び第2のエコーキャンセラ30A,30Bで構成される。   As shown in FIG. 1, the present embodiment includes a microphone 1, a speaker 2, a 2-wire to 4-wire conversion circuit 3, a reception signal amplifier VG, a voice switch 10, and first and second echo cancellers 30A and 30B. .

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帰還経路(音響エコー経路)HAC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2への入力信号)から推定した擬似エコー成分(音響エコー)を減算器32Aによりマイクロホン1の出力信号から減算することで音響エコーを抑制するものである。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相手の通話端末(例えば、インターホンシステムのドアホン子器など)におけるスピーカ−マイクロホン間の音響結合とにより形成される帰還経路(回線エコー経路)HLIN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(2線−4線変換回路3への入力信号、すなわち送話信号)から推定した擬似エコー成分(回線エコー)を減算器32Bにより受話信号から減算することで回線エコーを抑制するものである。 The first echo canceller 30A includes a well-known structure composed of the adaptive filter 31A and a subtractor 32A, the impulse response of the feedback path (acoustic echo path) H AC formed by the acoustic coupling between the speaker 2 microphone 1 The acoustic echo is suppressed by subtracting the pseudo echo component (acoustic echo) that is adaptively identified by the adaptive filter 31A and estimated from the reference signal (input signal to the speaker 2) from the output signal of the microphone 1 by the subtractor 32A. Is. The second echo canceller 30B also has a conventionally well-known configuration including an adaptive filter 31B and a subtractor 32B. The reflection due to impedance mismatch between the 2-wire-to-wire conversion circuit 3 and the transmission path and the counterpart The impulse response of the feedback path (line echo path) H LIN formed by the acoustic coupling between the speaker and the microphone in the telephone terminal (for example, intercom system door phone slave unit) is adaptively identified by the adaptive filter 31B and referred to The line echo is suppressed by subtracting the pseudo echo component (line echo) estimated from the signal (input signal to the 2-wire to 4-wire conversion circuit 3, that is, transmission signal) from the reception signal by the subtractor 32B. .

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部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部12と、送話側及び受話側の各損失挿入部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、受話側損失挿入部12の出力点Routから音響エコー経路HACを介して送話側損失挿入部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、送話側損失挿入部11の出力点Toutから回線エコー経路HLINを介して受話側損失挿入部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(送話側損失挿入部11の挿入損失量と受話側損失挿入部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損失挿入部11及び受話側損失挿入部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、本実施形態における第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は、DSP(Digital Signal Processor)のハードウェアをエコーキャンセラ用並びに音声スイッチ用のソフトウェア(プログラム)で制御することによって実現されている。従って、以下の説明における音声スイッチ10並びに第1及び第2のエコーキャンセラ30A,30Bの入出力信号(受話信号及び送話信号)は所定のサンプリング周期でサンプリングされ、且つA/D変換器により量子化されている。 The voice switch 10 includes a transmission side loss insertion unit 11 that inserts a loss into the signal path on the transmission side, a reception side loss insertion unit 12 that inserts a loss into the signal path on the reception side, and a transmission side and a reception side. And an insertion loss amount control unit 13 that controls the amount of loss inserted from each of the loss insertion units 11 and 12. The insertion loss amount control unit 13, the path to return to the input point Tin of the transmitting end losses insertion portion 11 from the output point Rout of the receiving-side loss insertion portion 12 via the acoustic echo path H AC (hereinafter, "sound side feedback path )) And a return path from the output point Tout of the transmission side loss insertion unit 11 to the input point Rin of the reception side loss insertion unit 12 via the line echo path H LIN (hereinafter referred to as “reception side feedback gain α”). (Referred to as “line-side feedback path”), and the loss amount to be inserted into the closed loop is estimated based on the estimated values α ′ and β ′ of the feedback gains α and β on the acoustic side and the line side. A total loss amount calculation unit 14 that calculates the sum (the sum of the insertion loss amount of the transmission side loss insertion unit 11 and the insertion loss amount of the reception side loss insertion unit 12), and the state of the call by monitoring the transmission signal and the reception signal According to the estimation result and the total loss calculation unit 14 It comprises an insertion loss amount distribution processing unit 15 that determines the distribution of each insertion loss amount of the side loss insertion unit 11 and the reception side loss insertion unit 12. The first and second echo cancellers 30A and 30B and the voice switch 10 according to the present embodiment control the DSP (Digital Signal Processor) hardware by software (programs) for the echo canceller and the voice switch. It has been realized. Therefore, the input / output signals (received signal and transmitted signal) of the voice switch 10 and the first and second echo cancellers 30A and 30B in the following description are sampled at a predetermined sampling period and quantized by the A / D converter. It has become.

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損失挿入部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損失挿入部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HACにて想定される最大遅延時間において受話側損失挿入部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損失挿入部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損失挿入部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損失挿入部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HLINにて想定される最大遅延時間において送話側損失挿入部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損失挿入部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。 The total loss amount calculation unit 14 estimates the time-average power of the input signal of the transmission side loss insertion unit 11 in a short time using a rectifier / smoothing device, a low-pass filter, and the like. estimating the time average power in a short time of the output signal of the receiving-side loss insertion portion 12 with a like, time of the output signal of the receiving-side loss insertion portion 12 in the maximum delay time assumed in acoustic side feedback path H AC A minimum value of the estimated value of the average power is obtained, and a value obtained by dividing the estimated value of the time average power of the input signal of the transmission side loss insertion unit 11 by this minimum value is set as the estimated value α ′ of the acoustic feedback gain α. The time average power of the input signal of the receiving side loss insertion unit 12 in a short time is estimated using a rectifying / smoothing device, a low-pass filter, etc. Insertion part 11 Estimating the time average power in a short time of the signal, determining the minimum value of the estimated value of the time average power of the output signal of the transmitter-side loss insertion unit 11 at the maximum delay time assumed in the line side feedback path H LIN, A value obtained by dividing the estimated value of the time average power of the input signal of the receiving side loss insertion unit 12 by this minimum value is defined as an 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.

挿入損失量分配処理部15では、送話側損失挿入部11の入出力信号及び受話側損失挿入部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を送話側損失挿入部11と受話側損失挿入部12に分配するように各損失挿入部11,12の挿入損失量を調整する。   The insertion loss amount distribution processing unit 15 monitors the input / output signals of the transmission side loss insertion unit 11 and the input / output signals of the reception side loss insertion unit 12, and compares the power levels of these signals and the presence / absence of a voice signal. The communication state (the reception state, the transmission state, etc.) is determined from the information of the information, and the total loss Lt is distributed to the transmission side loss insertion unit 11 and the reception side loss insertion unit 12 at a rate corresponding to the determined communication state. The insertion loss amount of each loss insertion part 11 and 12 is adjusted so that it may.

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、相手側の通話端末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。   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 and 30B are sufficiently provided from the start of a call with the other party's call terminal. It operates in the fixed mode during the period until convergence, and operates in the update mode during the period after the first and second echo cancellers 30A and 30B have sufficiently converged. 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.

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。   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.

ここで、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。   Here, the specific operation of the total loss amount calculation unit 14 in the update mode will be described with reference to the flowchart of FIG.

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)。   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).

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)。
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).

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、相手側の通話端末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。   Thus, 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 second echo cancellers 30A can be used just after the start of a call with the other party's call terminal. , 30B actively update the coefficient toward convergence, so that a sense of incongruity can be eliminated even when the acoustic feedback gain α and the line feedback gain β change significantly.

ところで、本実施形態では音声スイッチ10と第1のエコーキャンセラ30Aとの間の受話側の信号経路中に受話信号増幅器VGが設けられている。この受話信号増幅器VGは、本実施形態を使って相手の通話端末と通話している話者によって増幅度が手動で増減されるものであり、例えば、話者が聴力の低下した高齢者であっても受話信号増幅器VGの増幅度を増大して受話音量を大きくすることによって受話音声を聞き取りやすくしている。ここで、従来技術で述べたように受話信号増幅器VGの増幅度を増大した場合、受話音量が増大することで音響側帰還利得αが増えてしまうから、音響側帰還経路HAC並びに回線側帰還経路HLINにより形成される閉ループの一巡利得も増加してハウリングが生じやすくなってしまう。 By the way, in this embodiment, the reception signal amplifier VG is provided in the signal path on the reception side between the voice switch 10 and the first echo canceller 30A. In this reception signal amplifier VG, the amplification level is manually increased or decreased by a speaker who is talking to the other party's call terminal using this embodiment. For example, the speaker is an elderly person whose hearing ability has decreased. However, by increasing the amplification level of the reception signal amplifier VG and increasing the reception volume, the reception voice can be easily heard. Here, when increasing the amplification degree of the receiving signal amplifier VG as described in the prior art, because thereby increasing acoustic side feedback gain α by receiver volume is increased, the acoustic side feedback path H AC and the line side feedback The round loop gain formed by the path H LIN also increases, and howling is likely to occur.

そこで本実施形態では、話者が手動で受話信号増幅器VGの増幅度を増大して受話音量を大きくした場合、音声スイッチ10の総損失量算出部14において、受話信号増幅器VGの増幅度の増大分に略等しい量だけ総損失量を増やすようにしている。その結果、受話信号増幅器VGの増幅度を増大しても音響側帰還経路HAC並びに回線側帰還経路HLINにより形成される閉ループの一巡利得の増加が抑制されるので、ハウリングの発生を防止することができる。ここで、受話信号増幅器VGの増幅度は通話が終了する毎に初期値に戻るようになっており、したがって、総損失量算出部14が受話信号増幅器VGの増幅度の増大分に略等しい量だけ総損失量を増やすのは、話者が手動で受話信号増幅器VGの増幅度を増大して受話音量を大きくした当該通話中のみであって、次回の通話時には上述した固定モードで動作する。 Therefore, in this embodiment, when the speaker manually increases the amplification level of the reception signal amplifier VG to increase the reception volume, the total loss amount calculation unit 14 of the voice switch 10 increases the amplification level of the reception signal amplifier VG. The total loss is increased by an amount approximately equal to the minute. As a result, even if the amplification factor of the reception signal amplifier VG is increased, an increase in the closed loop loop gain formed by the acoustic side feedback path H AC and the line side feedback path H LIN is suppressed, thereby preventing howling. be able to. Here, the amplification degree of the reception signal amplifier VG returns to the initial value every time a call is finished, and therefore the total loss calculation unit 14 is an amount substantially equal to the increase in the amplification degree of the reception signal amplifier VG. The total loss amount is increased only during the call in which the speaker manually increases the amplification level of the reception signal amplifier VG to increase the reception volume, and operates in the above-described fixed mode at the next call.

なお、話者が手動で受話信号増幅器VGの増幅度を増大して受話音量を大きくした場合であっても、総損失量算出部14が受話信号増幅器VGの増幅度の増大分に略等しい量だけ総損失量を増やすことで回線側帰還利得βはほとんど影響を受けないので、第2のエコーキャンセラ30Bの適応フィルタ31Bにおいては、受話信号増幅器VGの増幅度が増大した後も増大前の擬似エコー成分の推定結果をそのまま用いればよい。   Even when the speaker manually increases the amplification level of the reception signal amplifier VG to increase the reception volume, the total loss amount calculation unit 14 is substantially equal to the increase in the amplification level of the reception signal amplifier VG. Since the line-side feedback gain β is hardly affected by increasing the total loss amount only, the adaptive filter 31B of the second echo canceller 30B is a pseudo signal before the increase even after the amplification factor of the reception signal amplifier VG is increased. The estimation result of the echo component may be used as it is.

また、受話信号増幅器VGが増幅度を増大又は減少させる際には、所定の時定数に従って緩やかに増幅度を変化させることが望ましい。つまり、受話信号増幅器VGの増幅度(受話音量)が急激に変化すると音声スイッチ10の総損失量算出部14において音響側帰還利得αの推定値α’を求める処理に過大な負荷がかかってしまうが、増幅度を緩やかに変化させることで推定値α’を求める処理にかかる負荷が軽減できる。ここで、第1のエコーキャンセラ30Aが具備する適応フィルタ31AはディジタルのFIRフィルタにより構成されており、擬似エコー成分の減算で消去されなかった消去誤差を最小とするように動作するアルゴリズムによってフィルタ係数を逐次修正している。したがって、第1のエコーキャンセラ30Aの適応フィルタ31Aに対し、増幅度を緩やかに変化させる際の時定数を、逐次修正するフィルタ係数が発散しないような値に設定すればよく、例えば、時定数の値を変えながらフィルタ係数の発散を確認する実験を行って適切な値を決めればよい。   Further, when the reception signal amplifier VG increases or decreases the amplification degree, it is desirable to change the amplification degree gently according to a predetermined time constant. That is, when the amplification degree (reception volume) of the reception signal amplifier VG changes abruptly, an excessive load is applied to the process of obtaining the estimated value α ′ of the acoustic feedback gain α in the total loss amount calculation unit 14 of the voice switch 10. However, it is possible to reduce the load on the process for obtaining the estimated value α ′ by gradually changing the amplification degree. Here, the adaptive filter 31A included in the first echo canceller 30A is composed of a digital FIR filter, and the filter coefficient is determined by an algorithm that operates to minimize the erasure error that has not been eliminated by subtraction of the pseudo echo component. Are being corrected sequentially. Therefore, for the adaptive filter 31A of the first echo canceller 30A, the time constant when the amplification degree is gradually changed may be set to a value such that the filter coefficient to be sequentially corrected does not diverge. An appropriate value may be determined by performing an experiment to confirm the divergence of the filter coefficient while changing the value.

本発明の実施形態を示すブロック図である。It is a block diagram which shows embodiment of this invention. 同上における音声スイッチの動作説明用のフローチャートである。It is a flowchart for operation | movement description of a voice switch in the same as the above.

符号の説明Explanation of symbols

1 マイクロホン
2 スピーカ
10 音声スイッチ
11 送話側損失挿入部
12 受話側損失挿入部
13 挿入損失量制御部
14 総損失量算出部
30A 第1のエコーキャンセラ
30B 第2のエコーキャンセラ
VG 受話信号増幅器
DESCRIPTION OF SYMBOLS 1 Microphone 2 Speaker 10 Voice switch 11 Transmission side loss insertion part 12 Reception side loss insertion part 13 Insertion loss amount control part 14 Total loss amount calculation part 30A 1st echo canceller 30B 2nd echo canceller VG Received signal amplifier

Claims (3)

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、マイクロホンとスピーカの音響結合によって生じる音響エコーを消去する第1のエコーキャンセラと、相手側の通話端末における音響結合や相手側の通話端末との間の回線における信号の回り込みによって生じる回線エコーを消去する第2のエコーキャンセラと、第1及び第2のエコーキャンセラの間に設けられ、音響エコー経路並びに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、音声スイッチと第1のエコーキャンセラの間に設けられ、スピーカへ出力される受話信号を増幅し且つ増幅度が可変である受話信号増幅手段とを備え、
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し、挿入損失量制御手段は、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とともに、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を算出する総損失量算出部と、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の算出値に応じて送話側損失挿入手段及び受話側挿入損失手段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とからなり、
総損失量算出部は、受話信号増幅手段の増幅度が増大した場合に当該増幅度の増大分に略等しい量だけ総損失量を増やすことを特徴とする拡声通話装置。
A microphone that outputs collected sound as a transmission signal, a speaker that rings in response to a reception signal from a call terminal on the other side, and a first echo canceller that eliminates acoustic echo generated by acoustic coupling between the microphone and the speaker; A second echo canceler for canceling line echo caused by acoustic coupling at the other party's call terminal and signal wraparound on the line with the other party's call terminal, and between the first and second echo cancellers A voice switch that suppresses howling by reducing a closed loop loop gain formed by the acoustic echo path and the line echo path, and a reception signal that is provided between the voice switch and the first echo canceller and is output to the speaker Receiving signal amplifying means with a variable amplification degree,
The voice switch includes transmission side loss insertion means for inserting loss into the signal path on the transmission side, reception side loss insertion means for inserting loss into the signal path on the reception side, and insertion of each loss on the transmission side and reception side Insertion loss amount control means for controlling the amount of loss inserted from the means, the insertion loss amount control means, the input point of the transmission side loss insertion means from the output point of the reception side loss insertion means via the acoustic echo path And estimate the line-side feedback gain of the path that returns from the output point of the transmission-side loss insertion means to the input point of the reception-side loss insertion means via the line echo path. A total loss amount calculation unit for calculating the sum of loss amounts to be inserted into the closed loop based on the estimated values of the feedback gains on the acoustic side and the line side, and the call state is estimated by monitoring the transmission signal and the reception signal. , This estimation result and the total loss calculation part Consists insertion loss amount distribution processing section for determining the distribution of the insertion loss of the transmitting end losses insertion means and the receiving side insertion loss means in accordance with output values,
The total loss amount calculation unit increases the total loss amount by an amount substantially equal to the increase in the amplification level when the amplification level of the received signal amplification means increases.
第2のエコーキャンセラは、回線エコー経路のインパルス応答を適応的に同定して当該回線エコー経路への入力信号から擬似エコー成分を推定する適応フィルタと、適応フィルタで推定された擬似エコー成分を回線エコー経路からの出力信号より減算する減算器とを具備し、適応フィルタは、受話信号増幅手段の増幅度が増大した後も増大前の擬似エコー成分の推定結果を用いることを特徴とする請求項1記載の拡声通話装置。   The second echo canceller adaptively identifies the impulse response of the line echo path and estimates the pseudo echo component from the input signal to the line echo path, and the line of the pseudo echo component estimated by the adaptive filter. The subtractor for subtracting from the output signal from the echo path, and the adaptive filter uses the estimation result of the pseudo echo component before the increase even after the amplification degree of the received signal amplification means is increased. The loudspeaker apparatus according to 1. 受話信号増幅手段は、所定の時定数に従って緩やかに増幅度を変化させることを特徴とする請求項1又は2記載の拡声通話装置。   The loudspeaker apparatus according to claim 1 or 2, wherein the reception signal amplifying means gently changes the amplification degree according to a predetermined time constant.
JP2005312003A 2005-10-26 2005-10-26 Loudspeaker call device Pending JP2007124162A (en)

Priority Applications (1)

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JP2005312003A JP2007124162A (en) 2005-10-26 2005-10-26 Loudspeaker call device

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JP2005312003A JP2007124162A (en) 2005-10-26 2005-10-26 Loudspeaker call device

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Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2019193068A (en) * 2018-04-24 2019-10-31 パナソニックIpマネジメント株式会社 Intercom device, intercom system, sound volume control method, and program

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2019193068A (en) * 2018-04-24 2019-10-31 パナソニックIpマネジメント株式会社 Intercom device, intercom system, sound volume control method, and program

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