JP4349296B2 - Loudspeaker - Google Patents

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JP4349296B2
JP4349296B2 JP2005030869A JP2005030869A JP4349296B2 JP 4349296 B2 JP4349296 B2 JP 4349296B2 JP 2005030869 A JP2005030869 A JP 2005030869A JP 2005030869 A JP2005030869 A JP 2005030869A JP 4349296 B2 JP4349296 B2 JP 4349296B2
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signal
closed loop
unit
ambient noise
level
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JP2005160120A (en
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実 福島
博昭 竹山
昌生 多氣
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Panasonic Corp
Panasonic Electric Works Co Ltd
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Matsushita Electric Works Ltd
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Description

本発明は、家庭内、ビルディング、工場等で用いられる拡声通話機に関するものである。   The present invention relates to a loudspeaker used in homes, buildings, factories, and the like.

従来より、インターホンや電話機あるいはPHS等の拡声通話機においては、スピーカからマイクロホンへの音響フィードバックおよびハイブリッド回路(2−4線変換回路)におけるインピーダンスの不整合により閉ループが形成され、増幅器の利得が大きすぎる等の理由により上記閉ループの利得が1倍以上になるとハウリングが生じるため、通話品質を確保する上でハウリングの抑圧が必要不可欠な課題となっていた。   Conventionally, in a loudspeaker such as an interphone, a telephone, or a PHS, a closed loop is formed by acoustic feedback from a speaker to a microphone and impedance mismatch in a hybrid circuit (2-4 wire conversion circuit), and the gain of the amplifier is large. Since howling occurs when the closed-loop gain becomes 1 or more for reasons such as too much, suppression of howling has become an indispensable issue in securing call quality.

そこで従来は、送受話信号のレべルに応じて受話信号または送話信号に所定量の損失を挿入することで閉ループ利得を抑圧するというハウリング抑圧方式が用いられてきた。また、別の方式としてエコーキャンセラを用いるものもあるが、エコーキャンセラにおける適応フィルタの係数が収束していない過渡状態や系の変動によりエコー経路が急激に変化した場合等において不安定化しやすいため、挿入損失と併用する場合が多い。   Therefore, conventionally, a howling suppression method has been used in which a closed loop gain is suppressed by inserting a predetermined amount of loss into a received signal or a transmitted signal in accordance with the level of the transmitted / received signal. In addition, there is another method that uses an echo canceller, but it tends to be unstable when the echo path changes suddenly due to a transient state where the coefficient of the adaptive filter in the echo canceller has not converged or system fluctuations. Often used in conjunction with insertion loss.

上記何れの場合においても、挿入損失量を大きくしすぎた場合には通話中に切断感を生じる(音声が途切れる)等の通話品質の劣化を招くため、挿入損失量を必要最小限とすることが望ましい。   In any of the above cases, if the amount of insertion loss is increased too much, the quality of the call will be degraded during a call (sound will be interrupted). Is desirable.

このような課題に対して本発明者らは既に、拡声通話系に形成される閉ループの利得余裕を随時監視し、推定された利得余裕値に基づいて挿入損失量を適応的に調整するようにした拡声通話機を提案している(特許文献1参照)。   In response to such problems, the present inventors have already monitored the gain margin of the closed loop formed in the voice communication system as needed, and adaptively adjust the insertion loss amount based on the estimated gain margin value. A loudspeaker has been proposed (see Patent Document 1).

図18(a)は上記出願に係る拡声通話機を示すブロック図であり、集音した音を送話側の音声信号(以下、送話信号と呼ぶ)として出力するマイクロホン1と、マイクロホン1からの送話信号を増幅する第1の増幅器2と、受話側の音声信号(以下、受話信号と呼ぶ)に応じて鳴動するスピーカ3と、スピーカ3へ出力される受話信号を増幅する第2の増幅器4と、送話側及び受話側と外部の通話回線との間で2−4線変換を行う2−4線変換手段たるハイブリッド回路5と、送話側及び受話側の信号経路に所定量の損失を挿入する損失挿入手段たる減衰器(アッテネータ)61,62と、各減衰器61,62の損失量を可変制御する制御器7と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン1からスピーカ3への音響結合及びハイブリッド回路5における反射により形成される閉ループにおける利得余裕を推定するとともに制御器7を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段8とを備えている。なお、挿入損失量調整手段8は送話側の信号経路上に設けることも可能である。 FIG. 18A is a block diagram showing a loudspeaker according to the above-mentioned application. From the microphone 1 that outputs the collected sound as a voice signal on the transmission side (hereinafter referred to as a transmission signal), A first amplifier 2 that amplifies the transmitted signal, a speaker 3 that rings in response to a voice signal on the receiving side (hereinafter referred to as a received signal), and a second that amplifies the received signal output to the speaker 3. A predetermined amount in the amplifier 4, the hybrid circuit 5 as 2-4 line conversion means for performing 2-4 line conversion between the transmission side and reception side and the external communication line, and the signal path on the transmission side and reception side Attenuators (attenuators) 6 1 , 6 2 , a controller 7 for variably controlling the loss amounts of the attenuators 6 1 , 6 2 , and a response to the sample signal sent to the signal path Depending on the signal, microphone 1 to speaker 3 Insertion loss amount adjusting means 8 is provided for estimating a gain margin in a closed loop formed by acoustic coupling to and reflection in the hybrid circuit 5 and adjusting a loss amount based on an estimation result via a controller 7. The insertion loss amount adjusting means 8 can be provided on the signal path on the transmission side.

一方、同図(b)は上記挿入損失量調整手段8の具体的な構成を示すブロック図であり、受話側の信号経路にサンプル信号たるインパルス信号を送出するインパルス信号発生器9と、受話信号にインパルス信号を加算する加算器10と、インパルス信号に対する応答信号の包絡線を検波する包絡線検波器11と、包絡線検波器11の出力信号に基づいて閉ループ系における利得余裕を推定する閉ループ利得余裕推定部12とを備えている。ここで、閉ループ利得余裕推定部12は、包絡線検波器11の出力信号レベルを予め求めた閾値レベルと比較する比較器13と、比較結果に基づいて閉ループ系の利得余裕値を推定する判定部14とを具備している。なお、サンプル信号としてはパルス幅が充分に短い単一パルス信号であってもよいし、バースト信号(バーストノイズ)であってもよい。そして、図19に示すように、第2の増幅器4→スピーカ3→マイクロホン1→第1の増幅器2→減衰器61→ハイブリッド回路5→減衰器62→挿入損失量調整手段8→第2の増幅器4により閉ループが形成されている。 On the other hand, FIG. 4B is a block diagram showing a specific configuration of the insertion loss amount adjusting means 8, and an impulse signal generator 9 for sending an impulse signal as a sample signal to the signal path on the reception side, and the reception signal An adder 10 for adding the impulse signal to the envelope, an envelope detector 11 for detecting the envelope of the response signal to the impulse signal, and a closed loop gain for estimating the gain margin in the closed loop system based on the output signal of the envelope detector 11 And a margin estimation unit 12. Here, the closed loop gain margin estimation unit 12 compares the output signal level of the envelope detector 11 with a predetermined threshold level, and a determination unit that estimates the gain margin value of the closed loop system based on the comparison result. 14. The sample signal may be a single pulse signal with a sufficiently short pulse width or a burst signal (burst noise). Then, as shown in FIG. 19, second amplifier 4 → speaker 3 → microphone 1 → first amplifier 2 → attenuator 6 1 → hybrid circuit 5 → attenuator 6 2 → insertion loss amount adjusting means 8 → second A closed loop is formed by the amplifier 4.

而して、加算器10の出力信号は、上記閉ループ系にサンプル信号を入力したときの応答信号であり、加算器10の出力信号を観測することにより閉ループ系のインパルス応答を推定することができる。抽出された包絡線成分は、閉ループ系の伝達関数における支配極(複素平面上において最も虚軸に近い極)の実部と密接な関係があり、包絡線の時間軸に対する減衰特性から閉ループ系の安定度すなわち利得余裕を推定することができる。しかしながら、図18(a)における加算器10の出力信号は、サンプル信号に対する応答成分の他に送受話音声信号および周囲雑音が重畳しており、上記挿入損失量調整手段8にはサンプル信号に対する応答成分のみを抽出する手段がないため、有音声区間および周囲騒音信号のレベルが大きい場合には利得余裕値の推定精度が劣化するという問題がある。このうち送受話音声による推定精度の劣化に対しては、受話信号のレベルから(あるいは送話信号のレベルであってもよい)、通話中の無音区間を検出する無音検出器を設け、この無音検出器にて無音区間が検出された場合にのみ閉ループ系にサンプル信号(インパルス信号又はバーストノイズ)を入力し、閉ループ利得余裕推定部12にて利得余裕値の推定処理を行うことにより推定精度を改善することができる。
特開平10−257159号公報
Thus, the output signal of the adder 10 is a response signal when a sample signal is input to the closed loop system, and the impulse response of the closed loop system can be estimated by observing the output signal of the adder 10. . The extracted envelope component is closely related to the real part of the dominant pole (the pole closest to the imaginary axis on the complex plane) in the transfer function of the closed-loop system. Stability, that is, gain margin can be estimated. However, in the output signal of the adder 10 in FIG. 18A, the transmission / reception voice signal and the ambient noise are superimposed in addition to the response component for the sample signal, and the insertion loss amount adjusting means 8 has a response to the sample signal. Since there is no means for extracting only the components, there is a problem that the estimation accuracy of the gain margin value deteriorates when the level of the voiced section and the ambient noise signal is large. Among these, for the deterioration of the estimation accuracy due to the transmitted / received speech, a silence detector for detecting a silent section during a call is provided from the level of the received signal (or may be the level of the transmitted signal). Only when a silent section is detected by the detector, a sample signal (impulse signal or burst noise) is input to the closed-loop system, and the closed-loop gain margin estimation unit 12 performs an estimation process of the gain margin value to increase the estimation accuracy. Can be improved.
Japanese Patent Laid-Open No. 10-257159

ところが、上記拡声通話系の閉ループ中に定常的な周囲騒音信号が存在し、かつ、そのレベルが大きい場合には、上記出願に係る拡声通話機では、利得余裕値を精度よく推定することができない。また、上記手法では、閉ループ中に音声信号が存在している状態では利得余裕値の推定処理を止める必要があるため、通話中に系の利得余裕値が変動するような場合にはこれに対応できず、系が不安定化する場合がある。   However, when a steady ambient noise signal exists in the closed loop of the above-mentioned loudspeaker communication system and its level is large, the loudspeaker according to the above application cannot accurately estimate the gain margin value. . In addition, in the above method, since it is necessary to stop the gain margin value estimation process when a voice signal is present in the closed loop, this is handled when the gain margin value of the system fluctuates during a call. It may not be possible and the system may become unstable.

本発明は上記問題に鑑みて為されたものであり、その目的とするところは、閉ループ中に周囲騒音や音声信号が存在し、かつそれらの信号のレベルが大きい場合においても十分な精度で利得余裕値を推定し、常に安定した通話を実現することのできる拡声通話機を提供することにある。   The present invention has been made in view of the above problems, and its object is to obtain a gain with sufficient accuracy even when ambient noise and audio signals are present in a closed loop and the levels of those signals are large. An object of the present invention is to provide a loudspeaker capable of estimating a margin value and always realizing a stable call.

請求項1の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する信号の自己相関関数を算出する自己相関関数算出部とを具備し、同期制御部は、自己相関関数の変数である時間シフト量と、所定値以上の雑音成分の抑圧量を得るために必要な同期加算回数との関係を、自己相関関数算出部により算出される自己相関関数と照合することにより、同期加算処理に要する時間を最小化するために必要なサンプル信号の送出間隔及び同期加算回数を求めて当該送出間隔及び同期加算回数を調整して成ることを特徴とする。 In order to achieve the above object, the first aspect of the present invention provides a microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And a control means for variably controlling the amount of loss inserted from the loss insertion means, and a control means for estimating a gain margin in a closed loop formed through a microphone and a speaker according to a response signal to the sample signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the insertion loss, and the insertion loss amount adjusting means sends a sample signal to the signal path. A sample signal generator, an adder for adding the sample signal to the audio signal, an envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path, and an envelope A closed loop gain margin estimation unit that estimates a gain margin in the closed loop based on the output signal of the line detector, and a timing control for sending the sample signal multiple times from the sample signal generator to the signal path at predetermined time intervals A synchronization control unit, a synchronization addition averaging unit that is provided in a preceding stage of the envelope detector and that receives timing information from the synchronization control unit and performs a synchronous addition averaging process on the response signal with respect to the sample signal; and a self-of-signal that exists in the closed loop ; and a self-correlation function calculation section for calculating a correlation function, the synchronization control unit comprises a time shift amount is a variable of the autocorrelation function, a predetermined value In order to minimize the time required for the synchronous addition process by collating the relationship with the number of times of synchronous addition necessary to obtain the suppression amount of the above noise component with the autocorrelation function calculated by the autocorrelation function calculation unit The transmission interval and the number of synchronous additions necessary for the sample signal are obtained, and the transmission interval and the number of synchronous additions are adjusted .

請求項の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする。 In order to achieve the above object, the invention according to claim 2 is a microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And a control means for variably controlling the amount of loss inserted from the loss insertion means, and a control means for estimating a gain margin in a closed loop formed through a microphone and a speaker according to a response signal to the sample signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the insertion loss, and the insertion loss amount adjusting means sends a sample signal to the signal path. A sample signal generator, an adder for adding the sample signal to the audio signal, an envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path, and an envelope A closed loop gain margin estimation unit that estimates a gain margin in the closed loop based on the output signal of the line detector, and a timing control for sending the sample signal multiple times from the sample signal generator to the signal path at predetermined time intervals A synchronization control unit, a synchronization addition averaging unit that is provided in a preceding stage of the envelope detector and receives the timing information from the synchronization control unit and performs a synchronous addition averaging process on the response signal with respect to the sample signal; and an ambient noise existing in the closed loop The noise level estimation unit for estimating the level and the synchronous addition operation in the synchronous addition averaging unit based on the estimation result by the noise level estimation unit A synchronization addition number adjustment unit that adjusts the noise level, and the synchronization addition number adjustment unit is estimated by the noise level estimation unit based on the statistical properties of the ambient noise level and the number of synchronization additions that have been quantitatively examined in advance. It is characterized in that the minimum number of synchronous additions necessary for minimizing the delay time associated with the synchronous addition average is set for the ambient noise level to be generated .

請求項の発明は、請求項の発明において、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部を挿入損失量調整手段が具備し、抽出された周囲騒音信号から騒音レベル推定部が閉ループ中に存在する周囲騒音のレベルを推定して成ることを特徴とする。 According to a third aspect of the present invention, in the second aspect of the present invention, the insertion loss amount adjusting means includes an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and the noise is extracted from the extracted ambient noise signal. The level estimation unit estimates the level of ambient noise existing in the closed loop.

請求項の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいてサンプル信号の送出レベルを調整する送出レベル調整部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部とを具備し、騒音レベル推定部は、周囲騒音抽出部で抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定して成り、送出レベル調整部は、閉ループ中の周囲騒音レベルに対するサンプル信号送出レベルの比が、閉ループにサンプル信号が送出される以前の周囲騒音レベルに対するサンプル信号送出レベルの前記比に対して所定値以上となるようにサンプル信号の送出レベルを調整することを特徴とする。 In order to achieve the above object, a fourth aspect of the present invention provides a microphone that outputs the collected sound as a voice signal on the transmission side, a first amplification unit that amplifies the voice signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And a control means for variably controlling the amount of loss inserted from the loss insertion means, and a control means for estimating a gain margin in a closed loop formed through a microphone and a speaker according to a response signal to the sample signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the insertion loss, and the insertion loss amount adjusting means sends a sample signal to the signal path. A sample signal generator, an adder for adding the sample signal to the audio signal, an envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path, and an envelope A closed loop gain margin estimation unit that estimates a gain margin in the closed loop based on the output signal of the line detector, and a timing control for sending the sample signal multiple times from the sample signal generator to the signal path at predetermined time intervals A synchronization control unit, a synchronization addition averaging unit that is provided in a preceding stage of the envelope detector and receives the timing information from the synchronization control unit and performs a synchronous addition averaging process on the response signal with respect to the sample signal; and an ambient noise existing in the closed loop The noise level estimation unit that estimates the level and the sample signal transmission level is adjusted based on the estimation results from the noise level estimation unit A transmission level adjustment unit, and an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and the noise level estimation unit is configured to perform a closed loop from the ambient noise signal extracted by the ambient noise extraction unit. The transmission level adjustment unit is configured by estimating the level of ambient noise existing, and the ratio of the sample signal transmission level to the ambient noise level in the closed loop is set so that the sample signal transmission with respect to the ambient noise level before the sample signal is transmitted to the closed loop. The transmission level of the sample signal is adjusted so as to be a predetermined value or more with respect to the ratio of the levels .

請求項の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部と、抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいてサンプル信号の送出レベルを調整する送出レベル調整部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、送出レベル調整部は、閉ループ中の周囲騒音レベルに対するサンプル信号送出レベルの比が、閉ループにサンプル信号が送出される以前の周囲騒音レベルに対するサンプル信号送出レベルの前記比が所定値以上となるようにサンプル信号の送出レベルを調整して成り、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする。 In order to achieve the above object, a fifth aspect of the present invention provides a microphone that outputs a collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And a control means for variably controlling the amount of loss inserted from the loss insertion means, and a control means for estimating a gain margin in a closed loop formed through a microphone and a speaker according to a response signal to the sample signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the insertion loss, and the insertion loss amount adjusting means sends a sample signal to the signal path. A sample signal generator, an adder for adding the sample signal to the audio signal, an envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path, and an envelope A closed loop gain margin estimation unit that estimates a gain margin in the closed loop based on the output signal of the line detector, and a timing control for sending the sample signal multiple times from the sample signal generator to the signal path at predetermined time intervals A synchronization control unit, a synchronization addition averaging unit that is provided in a preceding stage of the envelope detector and receives timing information from the synchronization control unit and performs a synchronous addition averaging process on the response signal with respect to the sample signal, and a signal transmitted during the closed loop Ambient noise extraction unit that extracts ambient noise signals, and estimates the level of ambient noise that exists in the closed loop from the extracted ambient noise signals A noise level estimation unit, a transmission level adjustment unit that adjusts the transmission level of the sample signal based on the estimation result by the noise level estimation unit, and the number of synchronization additions in the synchronization addition averaging unit based on the estimation result by the noise level estimation unit And a sending level adjusting unit that sends the sample signal to the ambient noise level before the sample signal is sent to the closed loop, with the ratio of the sample signal sending level to the ambient noise level in the closed loop being adjusted. The level of the sample signal is adjusted so that the level ratio is equal to or greater than a predetermined value. The synchronization addition number adjustment unit is a statistical property of the ambient noise level and the number of synchronization additions that are quantitatively examined in advance. In order to minimize the delay time associated with the synchronous averaging with respect to the ambient noise level estimated by the noise level estimation unit It is characterized in that the minimum number of synchronous additions required is set .

請求項の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する信号の自己相関関数を算出する自己相関関数算出部とを具備し、同期制御部は、自己相関関数の変数である時間シフト量と、所定値以上の雑音成分の抑圧量を得るために必要な同期加算回数との関係を、自己相関関数算出部により算出される自己相関関数と照合することにより、同期加算処理に要する時間を最小化するために必要な疑似白色信号発生器からの疑似白色信号の送出間隔及び同期加算回数を求めて当該送出間隔及び同期加算回数を調整することを特徴とする。 In order to achieve the above object, the invention according to claim 6 is a microphone that outputs the collected sound as an audio signal on the transmission side, a first amplification means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And control means for variably controlling the amount of loss inserted from the loss insertion means, and estimating and controlling the gain margin in the closed loop formed through the microphone and the speaker according to the response signal to the pseudo white signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the means, the insertion loss amount adjusting means is a pseudo white with a finite period in the signal path A pseudo white signal generator for inserting a signal, an adder for adding the pseudo white signal to the audio signal, and the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path. A cross-correlation calculation unit that obtains a cross-correlation value and estimates a closed-loop impulse response based on the obtained cross-correlation value, and an envelope detector that detects an envelope of the closed-loop impulse response signal estimated by the cross-correlation calculation unit A closed-loop gain margin estimation unit that estimates a gain margin in the closed-loop based on an output signal of the envelope detector, and a pseudo-white signal is transmitted from the pseudo-white signal generator to the signal path a plurality of times at predetermined time intervals. A synchronization control unit that controls the timing of the signal and a cross-correlation calculation unit that receives timing information from the synchronization control unit that is provided in the front stage of the envelope detector. A synchronous addition averaging unit that performs synchronous addition averaging processing on the estimated closed-loop impulse response signal, and an autocorrelation function calculation unit that calculates an autocorrelation function of a signal existing in the closed loop. The correlation between the time shift amount, which is a variable of the correlation function, and the number of synchronous additions necessary to obtain a noise component suppression amount greater than or equal to a predetermined value is checked against the autocorrelation function calculated by the autocorrelation function calculation unit. To obtain the transmission interval and the number of synchronous additions of the pseudo white signal from the pseudo white signal generator necessary for minimizing the time required for the synchronous addition processing, and adjust the transmission interval and the number of synchronous additions. To do.

請求項の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする。 In order to achieve the above object, a seventh aspect of the present invention provides a microphone that outputs the collected sound as a voice signal on the transmission side, a first amplification means that amplifies the voice signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And control means for variably controlling the amount of loss inserted from the loss insertion means, and estimating and controlling the gain margin in the closed loop formed through the microphone and the speaker according to the response signal to the pseudo white signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the means, the insertion loss amount adjusting means is a pseudo white with a finite period in the signal path A pseudo white signal generator for inserting a signal, an adder for adding the pseudo white signal to the audio signal, and the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path. A cross-correlation calculation unit that obtains a cross-correlation value and estimates a closed-loop impulse response based on the obtained cross-correlation value, and an envelope detector that detects an envelope of the closed-loop impulse response signal estimated by the cross-correlation calculation unit A closed-loop gain margin estimation unit that estimates a gain margin in the closed-loop based on an output signal of the envelope detector, and a pseudo-white signal is transmitted from the pseudo-white signal generator to the signal path a plurality of times at predetermined time intervals. A synchronization control unit that controls the timing of the signal and a cross-correlation calculation unit that receives timing information from the synchronization control unit that is provided in the front stage of the envelope detector. Based on the estimation results from the noise level estimation unit, the synchronous addition averaging unit that performs synchronous addition averaging processing on the estimated closed loop impulse response signal, the noise level estimation unit that estimates the level of ambient noise existing in the closed loop A synchronization addition number adjustment unit that adjusts the number of synchronization additions in the addition averaging unit, and the synchronization addition number adjustment unit is based on the statistical properties of the ambient noise level and the number of synchronization additions that have been quantitatively examined in advance. The minimum number of synchronous additions necessary for minimizing the delay time associated with the synchronous addition average is set for the ambient noise level estimated by the noise level estimation unit .

請求項の発明は、請求項の発明において、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部を挿入損失量調整手段が具備し、抽出された周囲騒音信号から騒音レベル推定部が閉ループ中に存在する周囲騒音のレベルを推定して成ることを特徴とする。 The invention according to claim 8 is the invention according to claim 7 , wherein the insertion loss amount adjusting means includes an ambient noise extraction unit for extracting an ambient noise signal from the signal transmitted during the closed loop, and noise is extracted from the extracted ambient noise signal. The level estimation unit estimates the level of ambient noise existing in the closed loop .

請求項の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて疑似白色信号の送出レベルを調整する送出レベル調整部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部とを具備し、騒音レベル推定部は、周囲騒音抽出部で抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定して成り、送出レベル調整部は、閉ループ中の周囲騒音レベルに対する擬似白色信号送出レベルの比が、閉ループに擬似白色信号が送出される以前の周囲騒音レベルに対する擬似白色信号送出レベルの前記比に対して所定値以上となるように擬似白色信号の送出レベルを調整することを特徴とする。 In order to achieve the above object, a ninth aspect of the present invention provides a microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And control means for variably controlling the amount of loss inserted from the loss insertion means, and estimating and controlling the gain margin in the closed loop formed through the microphone and the speaker according to the response signal to the pseudo white signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the means, the insertion loss amount adjusting means is a pseudo white with a finite period in the signal path A pseudo white signal generator for inserting a signal, an adder for adding the pseudo white signal to the audio signal, and the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path. A cross-correlation calculation unit that obtains a cross-correlation value and estimates a closed-loop impulse response based on the obtained cross-correlation value, and an envelope detector that detects an envelope of the closed-loop impulse response signal estimated by the cross-correlation calculation unit A closed-loop gain margin estimation unit that estimates a gain margin in the closed-loop based on an output signal of the envelope detector, and a pseudo-white signal is transmitted from the pseudo-white signal generator to the signal path a plurality of times at predetermined time intervals. A synchronization control unit that controls the timing of the signal and a cross-correlation calculation unit that receives timing information from the synchronization control unit that is provided in the front stage of the envelope detector. Based on the estimation result by the noise level estimation unit, the noise level estimation unit for estimating the level of ambient noise existing in the closed loop, A white level signal transmission level adjustment unit, and a surrounding noise extraction unit for extracting an ambient noise signal from a signal transmitted during a closed loop. The noise level estimation unit is extracted by the ambient noise extraction unit. The level of ambient noise existing in the closed loop is estimated from the ambient noise signal, and the transmission level adjustment unit sends the ratio of the pseudo white signal transmission level to the ambient noise level in the closed loop, and the pseudo white signal is transmitted to the closed loop. The pseudo white signal transmission level is adjusted to be equal to or greater than a predetermined value with respect to the ratio of the pseudo white signal transmission level to the previous ambient noise level. It is characterized by adjusting .

請求項10の発明は、上記目的を達成するために、集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部と、抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて疑似白色信号の送出レベルを調整する送出レベル調整部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、送出レベル調整部は、閉ループ中の周囲騒音レベルに対するサンプル信号送出レベルの比が、閉ループにサンプル信号が送出される以前の周囲騒音レベルに対するサンプル信号送出レベルの前記比が所定値以上となるようにサンプル信号の送出レベルを調整して成り、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする。 In order to achieve the above object, a tenth aspect of the present invention provides a microphone that outputs the collected sound as a voice signal on the transmission side, a first amplification means that amplifies the voice signal from the microphone, A speaker that rings in response to an audio signal, a second amplifying unit that amplifies the audio signal output to the speaker, and a loss insertion unit that inserts a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side And control means for variably controlling the amount of loss inserted from the loss insertion means, and estimating and controlling the gain margin in the closed loop formed through the microphone and the speaker according to the response signal to the pseudo white signal sent to the signal path Insertion loss amount adjusting means for adjusting the loss amount based on the estimation result via the means, and the insertion loss amount adjusting means is a pseudo white having a finite period in the signal path. A pseudo white signal generator for inserting a signal, an adder for adding the pseudo white signal to the audio signal, and the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path. A cross-correlation calculation unit that obtains a cross-correlation value and estimates a closed-loop impulse response based on the obtained cross-correlation value, and an envelope detector that detects an envelope of the closed-loop impulse response signal estimated by the cross-correlation calculation unit A closed-loop gain margin estimation unit that estimates a gain margin in the closed-loop based on an output signal of the envelope detector, and a pseudo-white signal is transmitted from the pseudo-white signal generator to the signal path a plurality of times at predetermined time intervals. And a cross-correlation calculation unit that receives timing information from the synchronization control unit provided in the previous stage of the envelope detector. A synchronous addition averaging unit that performs synchronous addition averaging processing of the estimated closed-loop impulse response signal, an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and a closed loop from the extracted ambient noise signal A noise level estimator that estimates the level of ambient noise existing in the sound source, a transmission level adjuster that adjusts the transmission level of the pseudo white signal based on an estimation result by the noise level estimator, and an estimation result by the noise level estimator A synchronous addition number adjusting unit that adjusts the number of synchronous additions in the synchronous addition averaging unit, and the transmission level adjustment unit sends the sample signal to the closed loop with the ratio of the sample signal transmission level to the ambient noise level in the closed loop. So that the ratio of the sample signal transmission level to the previous ambient noise level is equal to or greater than a predetermined value. The synchronization addition frequency adjustment unit is configured by adjusting the transmission level, and the ambient noise level estimated by the noise level estimation unit based on the statistical characteristics of the ambient noise level and the synchronization addition frequency that have been quantitatively examined in advance. The minimum number of synchronous additions necessary for minimizing the delay time associated with the synchronous addition average is set .

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、サンプル信号を入力したときに観測される応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まる。しかも、同期制御部においてサンプル信号の送出タイミングを制御し、同期加算平均部でサンプル信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、さらに、事前に算出された閉ループ中に存在する信号の自己相関関数より、同期加算処理に要する時間を最小化するようにサンプル信号の送出間隔及び同期加算回数を調整することで利得余裕値の推定処理を高速化することができる。 According to the first aspect of the present invention, the acoustic coupling from the speaker to the microphone or the 2-4 line conversion used when the 2-4 line conversion needs to be performed between the transmission side and the reception side and the external communication line. For the closed loop system formed due to reflection due to impedance mismatch in the means, the gain margin value is estimated by the attenuation characteristic of the envelope component of the response signal observed when the sample signal is input, and the desired gain Since the amount of insertion loss is adjusted by calculating the excess and deficiency relative to the margin value, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality without causing howling Can be maintained, and the amount of insertion loss is not increased more than necessary, thereby increasing the possibility of realizing two-way simultaneous call performance. Moreover, the synchronization control unit controls the transmission timing of the sample signal, the synchronous addition averaging unit performs the moving average process synchronized with the sample signal, and the envelope detector detects the envelope component from the output signal of the synchronous addition average unit. Because it is extracted, even when the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced, the gain margin value is estimated with sufficient accuracy, and the insertion loss amount can be set to an appropriate value As a result, a stable call can be realized at all times, and the sample signal is transmitted so as to minimize the time required for the synchronous addition process from the autocorrelation function of the signal existing in the closed loop calculated in advance. The gain margin value estimation process can be speeded up by adjusting the interval and the number of synchronous additions.

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、サンプル信号を入力したときに観測される応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まる。しかも、同期制御部においてサンプル信号の送出タイミングを制御し、同期加算平均部でサンプル信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、さらに、閉ループ系に存在する周囲騒音のレベルに応じて同期加算回数を必要最小限に設定することができるため、同期加算処理に要する計算時間を最小化することができ、しかも、1回の利得余裕値の推定処理に要するサンプル信号の送出回数を最小化することにより、サンプル信号が受話信号に重畳して聞こえることによる聴覚上の不快感、違和感を軽減することができる。 According to the second aspect of the present invention, acoustic coupling from the speaker to the microphone, or 2-4 line conversion used when 2-4 line conversion needs to be performed between the transmission side and the reception side and the external communication line. For the closed loop system formed due to reflection due to impedance mismatch in the means, the gain margin value is estimated by the attenuation characteristic of the envelope component of the response signal observed when the sample signal is input, and the desired gain Since the amount of insertion loss is adjusted by calculating the excess and deficiency relative to the margin value, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality without causing howling Can be maintained, and the amount of insertion loss is not increased more than necessary, thereby increasing the possibility of realizing two-way simultaneous call performance. Moreover, the synchronization control unit controls the transmission timing of the sample signal, the synchronous addition averaging unit performs the moving average process synchronized with the sample signal, and the envelope detector detects the envelope component from the output signal of the synchronous addition average unit. Because it is extracted, even when the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced, the gain margin value is estimated with sufficient accuracy, and the insertion loss amount can be set to an appropriate value Therefore, it is possible to always realize a stable call, and furthermore, since the number of synchronous additions can be set to the minimum necessary according to the level of ambient noise existing in the closed loop system, the calculation required for the synchronous addition processing Time can be minimized, and the sample signal can be received by minimizing the number of times the sample signal is sent to estimate the gain margin once. Hearing on the discomfort caused by sounds superimposed on the issue, it is possible to reduce the discomfort.

請求項の発明によれば、請求項の発明の効果に加えて、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応してサンプル信号の送出レベルあるいは同期加算回数を調整するため、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。 According to the invention of claim 3 , in addition to the effect of the invention of claim 2 , even when the ambient noise level in the closed loop changes during the process of estimating the gain margin value, the sample signal is adapted to this. Therefore, even when the ambient noise level fluctuates, the gain margin value can be accurately estimated following this.

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、サンプル信号を入力したときに観測される応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まる。しかも、同期制御部においてサンプル信号の送出タイミングを制御し、同期加算平均部でサンプル信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、さらに、閉ループ系に送出するサンプル信号のレベルを周囲騒音レベルに対して充分な大きさに設定することにより、観測点において高いS/N比でサンプル信号に対する応答信号を得ることができ、利得余裕値の推定精度を向上させることができ、しかも、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応してサンプル信号の送出レベルあるいは同期加算回数を調整するため、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。 According to the fourth aspect of the present invention, the acoustic coupling from the speaker to the microphone, or the 2-4 line conversion used when the 2-4 line conversion needs to be performed between the transmission side and the reception side and the external communication line. For the closed loop system formed due to reflection due to impedance mismatch in the means, the gain margin value is estimated by the attenuation characteristic of the envelope component of the response signal observed when the sample signal is input, and the desired gain Since the amount of insertion loss is adjusted by calculating the excess and deficiency relative to the margin value, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality without causing howling Can be maintained, and the amount of insertion loss is not increased more than necessary, thereby increasing the possibility of realizing two-way simultaneous call performance. Moreover, the synchronization control unit controls the transmission timing of the sample signal, the synchronous addition averaging unit performs the moving average process synchronized with the sample signal, and the envelope detector detects the envelope component from the output signal of the synchronous addition average unit. Because it is extracted, even when the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced, the gain margin value is estimated with sufficient accuracy, and the insertion loss amount can be set to an appropriate value Thus, a stable call can be realized at all times. Further, by setting the level of the sample signal sent to the closed loop system to be sufficiently large with respect to the ambient noise level, a high S / N at the observation point. The response signal to the sample signal can be obtained with the ratio, and the accuracy of the gain margin value estimation can be improved. Even when the ambient noise level changes, the sample signal transmission level or the number of synchronous additions is adjusted to adjust to this, so even if the ambient noise level fluctuates, it will follow this. Thus, the gain margin value can be estimated with high accuracy.

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、サンプル信号を入力したときに観測される応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まる。しかも、同期制御部においてサンプル信号の送出タイミングを制御し、同期加算平均部でサンプル信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、さらに、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応してサンプル信号の送出レベル及び同期加算回数を調整するため、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。 According to the invention of claim 5 , acoustic coupling from the speaker to the microphone, or 2-4 line conversion used when 2-4 line conversion is required between the transmitting side and the receiving side and an external telephone line. For the closed loop system formed due to reflection due to impedance mismatch in the means, the gain margin value is estimated by the attenuation characteristic of the envelope component of the response signal observed when the sample signal is input, and the desired gain Since the amount of insertion loss is adjusted by calculating the excess and deficiency relative to the margin value, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality without causing howling Can be maintained, and the amount of insertion loss is not increased more than necessary, thereby increasing the possibility of realizing two-way simultaneous call performance. Moreover, the synchronization control unit controls the transmission timing of the sample signal, the synchronous addition averaging unit performs the moving average process synchronized with the sample signal, and the envelope detector detects the envelope component from the output signal of the synchronous addition average unit. Because it is extracted, even when the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced, the gain margin value is estimated with sufficient accuracy, and the insertion loss amount can be set to an appropriate value Therefore, even when the ambient noise level in the closed loop changes during the process of estimating the gain margin value, the sample signal transmission level is adapted to this. Since the number of synchronous additions is adjusted, even when the ambient noise level fluctuates, the gain margin value can be accurately estimated following this.

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、周期が有限の疑似白色信号を入力したときに観測される応答信号と当該疑似白色信号との相互相関値を求めるとともに求めた相互相関値から推定される閉ループ系のインパルス応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まり、しかも、利得余裕の推定に疑似白色信号を用いるので、使用者に聴感上の不快感を与えるのを防止することができ、さらに、同期制御部において疑似白色信号の送出タイミングを制御し、同期加算平均部で疑似白色信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、また、事前に算出された閉ループ中に存在する信号の自己相関関数より、同期加算処理に要する時間を最小化するように疑似白色信号の送出間隔及び同期加算回数を調整することで利得余裕値の推定処理を高速化することができる。 According to the sixth aspect of the present invention, the acoustic coupling from the speaker to the microphone, or the 2-4 line conversion used when the 2-4 line conversion needs to be performed between the transmission side and the reception side and the external communication line. The cross-correlation value between the response signal observed when a pseudo white signal with a finite period is input and the pseudo white signal is obtained for a closed loop system formed due to reflection due to impedance mismatch in the means. The gain margin value is estimated based on the attenuation characteristic of the envelope component of the impulse response signal of the closed loop system estimated from the obtained cross-correlation value, and the amount of insertion loss is adjusted by calculating the excess or deficiency relative to the desired gain margin value. Therefore, the gain margin of the closed loop system until howling can be maintained at a desired value, and stable call quality can be maintained without generating howling. In addition, since the amount of insertion loss is not increased more than necessary, the possibility of two-way simultaneous call performance is increased, and the pseudo white signal is used to estimate the gain margin, so that the user feels uncomfortable. In addition, the synchronous control unit controls the transmission timing of the pseudo white signal, the synchronous addition averaging unit performs the moving average process synchronized with the pseudo white signal, and the output of the synchronous addition average unit Envelope components are extracted from the signal by the envelope detector, so even if the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced and the gain margin value is estimated with sufficient accuracy Therefore, the amount of insertion loss can be set to an appropriate value, so that a stable call can always be realized, and the autocorrelation of the signal existing in the closed loop calculated in advance is possible. More, it is possible to speed up the process of estimating the gain margin value by adjusting the transmission interval and the synchronization addition number of the pseudo white signal to minimize the time required for the synchronous addition processing.

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、周期が有限の疑似白色信号を入力したときに観測される応答信号と当該疑似白色信号との相互相関値を求めるとともに求めた相互相関値から推定される閉ループ系のインパルス応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まり、しかも、利得余裕の推定に疑似白色信号を用いるので、使用者に聴感上の不快感を与えるのを防止することができ、さらに、同期制御部において疑似白色信号の送出タイミングを制御し、同期加算平均部で疑似白色信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、また、閉ループ系に存在する周囲騒音のレベルに応じて同期加算回数を必要最小限に設定することができるため、同期加算処理に要する計算時間を最小化することができ、さらに、1回の利得余裕値の推定処理に要する疑似白色信号の送出回数を最小化することにより、疑似白色信号が受話信号に重畳して聞こえることによる聴覚上の不快感、違和感を軽減することができる。 According to the invention of claim 7 , acoustic coupling from a speaker to a microphone, or 2-4 line conversion used when 2-4 line conversion needs to be performed between the transmitting side and the receiving side and an external telephone line. The cross-correlation value between the response signal observed when a pseudo white signal with a finite period is input and the pseudo white signal is obtained for a closed loop system formed due to reflection due to impedance mismatch in the means. The gain margin value is estimated based on the attenuation characteristic of the envelope component of the impulse response signal of the closed loop system estimated from the obtained cross-correlation value, and the amount of insertion loss is adjusted by calculating the excess or deficiency relative to the desired gain margin value. Therefore, the gain margin of the closed loop system until howling can be maintained at a desired value, and stable call quality can be maintained without generating howling. In addition, since the amount of insertion loss is not increased more than necessary, the possibility of two-way simultaneous call performance is increased, and the pseudo white signal is used to estimate the gain margin, so that the user feels uncomfortable. In addition, the synchronous control unit controls the transmission timing of the pseudo white signal, the synchronous addition averaging unit performs the moving average process synchronized with the pseudo white signal, and the output of the synchronous addition average unit Envelope components are extracted from the signal by the envelope detector, so even if the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced and the gain margin value is estimated with sufficient accuracy Therefore, the amount of insertion loss can be set to an appropriate value, so that stable calls can be realized at all times, and synchronous addition is performed according to the level of ambient noise existing in the closed loop system. Since the number can be set to the minimum necessary, the calculation time required for the synchronous addition process can be minimized, and the number of pseudo white signal transmissions required for one gain margin value estimation process can be minimized. By doing so, it is possible to reduce auditory discomfort and discomfort due to the fact that the pseudo white signal is superimposed on the received signal.

請求項の発明によれば、請求項の発明の効果に加えて、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応して疑似白色信号の送出レベルあるいは同期加算回数を調整するため、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。 According to the eighth aspect of the invention, in addition to the effect of the seventh aspect of the invention, even when the ambient noise level in the closed loop changes during the process of estimating the gain margin value, the pseudo white color is adapted to this. Since the signal transmission level or the number of synchronous additions is adjusted, even when the ambient noise level fluctuates, the gain margin value can be accurately estimated following this.

請求項の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、周期が有限の疑似白色信号を入力したときに観測される応答信号と当該疑似白色信号との相互相関値を求めるとともに求めた相互相関値から推定される閉ループ系のインパルス応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まり、しかも、利得余裕の推定に疑似白色信号を用いるので、使用者に聴感上の不快感を与えるのを防止することができ、さらに、同期制御部において疑似白色信号の送出タイミングを制御し、同期加算平均部で疑似白色信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、また、閉ループ系に送出する疑似白色信号のレベルを周囲騒音レベルに対して充分な大きさに設定することにより、観測点において高いS/N比で疑似白色信号に対する応答信号を得ることができ、利得余裕値の推定精度を向上させることができ、さらに、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応して疑似白色信号の送出レベルあるいは同期加算回数を調整するため、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。 According to the ninth aspect of the present invention, the acoustic coupling from the speaker to the microphone, or the 2-4 line conversion used when the 2-4 line conversion needs to be performed between the transmission side and the reception side and the external communication line. The cross-correlation value between the response signal observed when a pseudo white signal with a finite period is input and the pseudo white signal is obtained for a closed loop system formed due to reflection due to impedance mismatch in the means. The gain margin value is estimated based on the attenuation characteristic of the envelope component of the impulse response signal of the closed loop system estimated from the obtained cross-correlation value, and the amount of insertion loss is adjusted by calculating the excess or deficiency relative to the desired gain margin value. Therefore, the gain margin of the closed loop system until howling can be maintained at a desired value, and stable call quality can be maintained without generating howling. In addition, since the amount of insertion loss is not increased more than necessary, the possibility of two-way simultaneous call performance is increased, and the pseudo white signal is used to estimate the gain margin, so that the user feels uncomfortable. In addition, the synchronous control unit controls the transmission timing of the pseudo white signal, the synchronous addition averaging unit performs the moving average process synchronized with the pseudo white signal, and the output of the synchronous addition average unit Envelope components are extracted from the signal by the envelope detector, so even if the level of stationary ambient noise that exists in the closed loop is large, the effect of ambient noise is reduced and the gain margin value is estimated with sufficient accuracy Therefore, the amount of insertion loss can be set to an appropriate value, so that stable conversation can be realized at all times, and the level of the pseudo white signal sent to the closed loop system can be set to the ambient noise level. By setting a sufficient size for the signal, it is possible to obtain a response signal for the pseudo white signal with a high S / N ratio at the observation point, improve the estimation accuracy of the gain margin value, Even when the ambient noise level in the closed loop changes during the gain margin estimation process, the level of ambient noise fluctuates in order to adjust the sending level of the pseudo white signal or the number of synchronous additions in response to this. Even in such a case, the gain margin value can be accurately estimated following this.

請求項10の発明によれば、スピーカからマイクヘの音響結合や送話側及び受話側と外部の通話回線との間で2−4線変換を行う必要がある場合に用いられる2−4線変換手段におけるインピーダンスの不整合による反射が原因となり形成される閉ループ系に対して、周期が有限の疑似白色信号を入力したときに観測される応答信号と当該疑似白色信号との相互相関値を求めるとともに求めた相互相関値から推定される閉ループ系のインパルス応答信号の包絡線成分の減衰特性により利得余裕値を推定し、所望の利得余裕値に対する過不足分を算出することで挿入損失量を調整するため、ハウリングが発生するまでの閉ループ系の利得余裕を所望の値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができるとともに、必要以上に挿入損失量を大きくすることがないため、双方向同時通話性能の実現可能性が高まり、しかも、利得余裕の推定に疑似白色信号を用いるので、使用者に聴感上の不快感を与えるのを防止することができ、さらに、同期制御部において疑似白色信号の送出タイミングを制御し、同期加算平均部で疑似白色信号に同期した移動平均処理を行うとともに、同期加算平均部の出力信号から包絡線検波器にて包絡線成分を抽出するため、閉ループ中に存在する定常的な周囲騒音のレベルが大きい場合においても周囲騒音による影響を軽減して十分な精度で利得余裕値を推定し、挿入損失量が適切な値に設定可能となって、常に安定した通話を実現することができ、また、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応して疑似白色信号の送出レベル及び同期加算回数を調整するため、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。 According to the invention of claim 10 , acoustic coupling from a speaker to a microphone, or 2-4 line conversion used when 2-4 line conversion needs to be performed between a transmitting side and a receiving side and an external communication line. The cross-correlation value between the response signal observed when a pseudo white signal with a finite period is input and the pseudo white signal is obtained for a closed loop system formed due to reflection due to impedance mismatch in the means. The gain margin value is estimated based on the attenuation characteristic of the envelope component of the impulse response signal of the closed loop system estimated from the obtained cross-correlation value, and the amount of insertion loss is adjusted by calculating the excess or deficiency relative to the desired gain margin value. Therefore, the gain margin of the closed loop system until howling occurs can be maintained at a desired value, and stable call quality can be maintained without causing howling. In addition, since the amount of insertion loss is not increased more than necessary, the feasibility of two-way simultaneous call performance is increased, and the pseudo white signal is used to estimate the gain margin, so that the user is not audible. In addition, the synchronization control unit controls the transmission timing of the pseudo white signal, the synchronous addition averaging unit performs the moving average process synchronized with the pseudo white signal, and the synchronous addition averaging unit Since the envelope component is extracted from the output signal by the envelope detector, even when the level of stationary ambient noise existing in the closed loop is large, the effect of ambient noise is reduced and the gain margin value is obtained with sufficient accuracy. Estimated and insertion loss can be set to an appropriate value, so that a stable call can always be realized, and the ambient noise level during closed loop during the gain margin estimation process Even in such a case, the pseudo white signal transmission level and the number of synchronous additions are adjusted to adapt to this, so even if the ambient noise level fluctuates, the gain margin can be accurately tracked. The value can be estimated.

本発明の実施形態を説明する前に、基本となる特許文献1に記載された拡声通話機の構成並びに動作について詳細に説明する。   Before describing an embodiment of the present invention, the configuration and operation of a loudspeaker described in Patent Document 1 serving as a base will be described in detail.

従来技術でも説明したように、特許文献1に記載された拡声通話機は、集音した音を送話側の音声信号(以下、送話信号と呼ぶ)として出力するマイクロホン1と、マイクロホン1からの送話信号を増幅する第1の増幅器2と、受話側の音声信号(以下、受話信号と呼ぶ)に応じて鳴動するスピーカ3と、スピーカ3へ出力される受話信号を増幅する第2の増幅器4と、送話側及び受話側と外部の通話回線との間で2−4線変換を行う2−4線変換手段たるハイブリッド回路5と、送話側及び受話側の信号経路に所定量の損失を挿入する損失挿入手段たる減衰器(アッテネータ)61,62と、各減衰器61,62の損失量を可変制御する制御器7と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン1及びスピーカ3を通じて形成される閉ループ系における利得余裕を推定するとともに制御器7を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段8とを備えている。 As described in the prior art, the loudspeaker described in Patent Document 1 includes a microphone 1 that outputs a collected sound as a voice signal on the transmission side (hereinafter referred to as a transmission signal), and a microphone 1. A first amplifier 2 that amplifies the transmitted signal, a speaker 3 that rings in response to a voice signal on the receiving side (hereinafter referred to as a received signal), and a second that amplifies the received signal output to the speaker 3. A predetermined amount in the amplifier 4, the hybrid circuit 5 as 2-4 line conversion means for performing 2-4 line conversion between the transmission side and reception side and the external communication line, and the signal path on the transmission side and reception side Attenuators (attenuators) 6 1 , 6 2 , a controller 7 for variably controlling the loss amounts of the attenuators 6 1 , 6 2 , and a response to the sample signal sent to the signal path Depending on the signal, microphone 1 and speaker 3 and an insertion loss amount adjusting means 8 for adjusting the loss amount based on the estimation result via the controller 7 while estimating the gain margin in the closed-loop system formed through 3.

一方、挿入損失量調整手段8は、受話側の信号経路にサンプル信号たるインパルス信号を送出するインパルス信号発生器9と、受話信号にインパルス信号を加算する加算器10と、インパルス信号に対する応答信号の包絡線を検波する包絡線検波器11と、包絡線検波器11の出力信号に基づいて閉ループ系における利得余裕を推定する閉ループ利得余裕推定部12とを備え、マイコンやDSP(DigitalSignalProcessor)などで構成される。ここで、閉ループ利得余裕推定部12は、後述するように包絡線検波器11の出力信号レベルを予め求めた閾値レベルと比較する比較器13と、比較結果に基づいて閉ループ系の利得余裕値を推定する判定部14とを具備している。なお、サンプル信号として用いるインパルス信号は、パルス幅が充分に短い単一パルス信号であってもよい。また包絡線検波器11は、整流回路とローパスフィルタ回路の合成回路や、巡回型ローパスフィルタやリーク積分器等のデジタル回路、あるいはDSP(DigitalSignalProcessor)などの信号処理手段によって構成することができる。   On the other hand, the insertion loss amount adjusting means 8 includes an impulse signal generator 9 for sending an impulse signal as a sample signal to the signal path on the reception side, an adder 10 for adding the impulse signal to the reception signal, and a response signal for the impulse signal. An envelope detector 11 for detecting an envelope, and a closed loop gain margin estimation unit 12 for estimating a gain margin in a closed loop system based on an output signal of the envelope detector 11, are configured by a microcomputer, a DSP (Digital Signal Processor), or the like. Is done. Here, the closed loop gain margin estimation unit 12 compares the output signal level of the envelope detector 11 with a previously obtained threshold level as described later, and calculates the gain margin value of the closed loop system based on the comparison result. And a determination unit 14 for estimation. The impulse signal used as the sample signal may be a single pulse signal with a sufficiently short pulse width. The envelope detector 11 can be composed of a synthesis circuit of a rectifier circuit and a low-pass filter circuit, a digital circuit such as a cyclic low-pass filter and a leak integrator, or a signal processing means such as a DSP (Digital Signal Processor).

また、図19に示すように第2の増幅器4→スピーカ3→マイクロホン1→第1の増幅器2→減衰器61→ハイブリッド回路5→減衰器62→挿入損失量調整手段8→第2の増幅器4により閉ループが形成されている。ここで、各部における伝達関数を以下のように定義する。 Further, as shown in FIG. 19, the second amplifier 4 → speaker 3 → microphone 1 → first amplifier 2 → attenuator 6 1 → hybrid circuit 5 → attenuator 6 2 → insertion loss amount adjusting means 8 → second A closed loop is formed by the amplifier 4. Here, the transfer function in each part is defined as follows.

S :スピーカ3の電気機械変換特性
G :スピーカ3からマイクロホン1への音響伝達特性
M :マイクロホン1の音響電気変換特性
Kr:第2の増幅器4の増幅特性
Kx:第1の増幅器2の増幅特性
Ar:受話側の減衰器62の減衰特性
Ax:送話側の減衰器61の減衰特性
Γ :ハイブリッド回路5における反射伝達関数
また、インパルス信号発生器9から出力されるインパルス信号をP、外部の通話回線から伝送されてくる遠端話者音声入力信号をY(以下、「受話信号」と呼ぶ)、マイクロホン1の集音する近端話者音声信号と周囲雑音との和をX(以下、「音響信号」と呼ぶ)とすると、挿入損失量調整手段8の構成要素の一つである加算器10の出力信号(応答信号)Qは下記式で表される。
S: Electromechanical conversion characteristic of speaker 3 G: Acoustic transfer characteristic from speaker 3 to microphone 1 M: Acoustoelectric conversion characteristic of microphone 1 Kr: Amplification characteristic of second amplifier 4 Kx: Amplification characteristic of first amplifier 2 Ar: damping characteristics of the receiving side of the attenuator 6 2 Ax: attenuation characteristic of the attenuator 61 of the transmitter side gamma: reflection transfer function in the hybrid circuit 5 also an impulse signal outputted from the impulse signal generator 9 P, The far-end speaker voice input signal transmitted from the external telephone line is Y (hereinafter referred to as “received signal”), and the sum of the near-end talker voice signal collected by the microphone 1 and the ambient noise is X ( (Hereinafter referred to as “acoustic signal”), the output signal (response signal) Q of the adder 10 which is one of the components of the insertion loss amount adjusting means 8 is expressed by the following equation.

Figure 0004349296
Figure 0004349296

なお、上記式のL(s)は上記閉ループ系における一巡伝達関数、sはラプラス変数をそれぞれ表す。   In the above equation, L (s) represents a one-round transfer function in the closed loop system, and s represents a Laplace variable.

ここで、閉ループ系の安定性は上記一巡伝達関数L(s)により判別することができる。すなわち、極座標系における一巡伝達関数L(s)のθ成分(=∠L(s))が∠L(s)=2nπ(nは整数)となる全ての周波数において、r成分(=|L(s)|)が|L(s)|<1ならば閉ループ系は安定、|L(s)|≧1となる周波数が存在すれば閉ループ系は不安定となり、その周波数において発振してハウリングが生じる。また、閉ループ系が安定である場合に、∠L(s)=2nπとなる全ての周波数における利得の最大値をLMAXとすれば、閉ループ系の利得余裕値は1/LMAXで表される。よって、閉ループ系の安定性の尺度は閉ループ利得余裕値により評価することができる。 Here, the stability of the closed-loop system can be discriminated by the one-round transfer function L (s). That is, the r component (= | L () at all frequencies where the θ component (= ∠L (s)) of the circular transfer function L (s) in the polar coordinate system is ∠L (s) = 2nπ (n is an integer). If s) |) is | L (s) | <1, the closed loop system is stable, and if there is a frequency where | L (s) | ≧ 1, the closed loop system becomes unstable and oscillates at that frequency and howling occurs. Arise. Further, when the closed loop system is stable, if the maximum gain value at all frequencies where ∠L (s) = 2nπ is L MAX , the gain margin value of the closed loop system is expressed by 1 / L MAX. . Therefore, a measure of the stability of the closed loop system can be evaluated by the closed loop gain margin value.

一方、閉ループ利得余裕値は、閉ループ系のインパルス応答特性と密接な関係があり、閉ループ利得余裕値が大きいほどインパルス応答信号Qの振幅が時間とともに急激に減衰し、閉ループ利得余裕値が小さいほど減衰が緩やかになる。そこで、閉ループ利得余裕推定部12において閉ループ利得余裕値を推定するためのサンプル信号(インパルス信号)Pを上記閉ループ系に与えたときの応答信号Qを観測し、その応答信号Qの包絡線成分から閉ループ系の時定数に関する情報を抽出して閉ループ系におけるハウリング発生限界までの閉ループ利得余裕値の推定を行うとともに、推定結果に基づき、挿入損失量調整手段8にて上記所望の閉ループ利得余裕値に対する実際の挿入損失量の過不足分を算出して閉ループ系への挿入損失量を調整する。   On the other hand, the closed loop gain margin value has a close relationship with the impulse response characteristic of the closed loop system. The larger the closed loop gain margin value, the more rapidly the amplitude of the impulse response signal Q attenuates with time, and the smaller the closed loop gain margin value, the more attenuated. Becomes moderate. Accordingly, the closed loop gain margin estimation unit 12 observes the response signal Q when the sample signal (impulse signal) P for estimating the closed loop gain margin value is applied to the closed loop system, and the envelope signal component of the response signal Q is observed. Information on the time constant of the closed loop system is extracted to estimate the closed loop gain margin value up to the howling occurrence limit in the closed loop system, and based on the estimation result, the insertion loss amount adjusting means 8 is used to estimate the desired closed loop gain margin value. Calculate the amount of insertion loss to the closed loop system by calculating the excess and deficiency of the actual amount of insertion loss.

すなわち、上述のように包絡線検波器11で得られる応答信号Qの包絡線成分の時間特性が閉ループ利得余裕値が大きいほど減衰が早く且つ小さいほど減衰が緩やかになるという性質を有することから、閉ループ利得余裕推定部12において事前に学習された種々の利得余裕値に対する包絡線検波器11の出力データから閾値レベルを求めておき、比較器13において観測される包絡線検波器11の出力信号レベルを上記閾値レベルと比較することにより、その比較結果に基づいて判定部14にて閉ループ利得余裕値が推定できる。そして、その推定結果から、閉ループ利得余裕値を設計仕様で定めた値とするために必要な損失量を挿入するべく、制御器7に信号を伝送して制御器7によって減衰器61,62の減衰量を調節している。 That is, since the time characteristic of the envelope component of the response signal Q obtained by the envelope detector 11 as described above has the property that the larger the closed loop gain margin value, the faster the attenuation and the smaller the attenuation, the slower the attenuation. A threshold level is obtained from the output data of the envelope detector 11 for various gain margin values learned in advance by the closed loop gain margin estimation unit 12, and the output signal level of the envelope detector 11 observed by the comparator 13. Is compared with the threshold level, the determination unit 14 can estimate the closed-loop gain margin based on the comparison result. Then, from the estimation result, a signal is transmitted to the controller 7 and the attenuators 6 1 , 6 are transmitted by the controller 7 in order to insert a loss amount necessary for setting the closed loop gain margin value to a value determined by the design specifications. The attenuation of 2 is adjusted.

上述のように上記拡声通話機では、インパルス信号に対する応答信号から、閉ループ系での利得余裕値を推定し、この利得余裕値の推定値と、目標とする利得余裕値との差分を算出し、その算出結果に基づいて信号経路に挿入する損失量を調整する挿入損失量調整手段8を備えているので、通話中においても閉ループ系の安定度に応じて挿入損失量を制御し、常に利得余裕値を仕様で定めた値に維持することができ、ハウリングを生じることなく安定した通話品質を維持することができる。また、従来例に比較して必要以上に挿入損失量を大きくする必要がないため、双方向同時通話性能の実現可能性が高まるという利点もある。   As described above, in the loudspeaker, the gain margin value in the closed loop system is estimated from the response signal to the impulse signal, the difference between the estimated gain margin value and the target gain margin value is calculated, Since the insertion loss amount adjusting means 8 for adjusting the loss amount inserted into the signal path based on the calculation result is provided, the insertion loss amount is controlled according to the stability of the closed loop system even during a call, and the gain margin is always obtained. The value can be maintained at a value determined by the specification, and stable call quality can be maintained without causing howling. Moreover, since it is not necessary to increase the amount of insertion loss more than necessary as compared with the conventional example, there is an advantage that the possibility of realizing bidirectional simultaneous call performance is increased.

なお、上記基本構成並びに後述する各実施形態においてはサンプル信号発生器にインパルス信号発生器9を用いたが、代わりにバースト信号発生器を用いてバースト信号をサンプル信号に用いてもよい。この場合には、信号を送出している状態から送出を停止した瞬間からの閉ループ系の過渡応答を観測し、その包絡線成分から閉ループ系の時定数に関する情報を抽出する。而してサンプル信号にバースト信号を用いた場合にも、信号送出時間が閉ループ系の遅延時間に対して十分に短ければ、その応答波形の包絡線成分より閉ループ系の時定数に関する情報を抽出することができ、バースト信号の信号送出時間を適切な値とすることにより通話中における違和感をなくし、サンプル信号の送出による通話品質の劣化を抑えることができるという利点がある。但し、種々の利得余裕値に対する閾値レべルを求めておく必要があることはインパルス信号の場合と同様である。   Although the impulse signal generator 9 is used as the sample signal generator in the basic configuration and each of the embodiments described later, a burst signal may be used as the sample signal by using a burst signal generator instead. In this case, the transient response of the closed loop system from the moment when the transmission is stopped from the state in which the signal is transmitted is observed, and information on the time constant of the closed loop system is extracted from the envelope component. Thus, even when a burst signal is used as the sample signal, if the signal transmission time is sufficiently shorter than the delay time of the closed loop system, information on the time constant of the closed loop system is extracted from the envelope component of the response waveform. In addition, there is an advantage that by setting the signal transmission time of the burst signal to an appropriate value, it is possible to eliminate a sense of incongruity during the call and to suppress deterioration in call quality due to the transmission of the sample signal. However, as in the case of the impulse signal, it is necessary to obtain threshold levels for various gain margin values.

ところで、上記基本構成においては式1で表される応答信号Qから利得余裕値を推定しているが、上記式1からも明らかなように応答信号Qにはサンプル信号Pのみに対する応答信号(式1の右辺第1項)以外の信号が含まれている。そのため、受話信号Y及びマイクロホン1の集音する音響信号Xのレベルが大きく、式1の右辺第2項及び第3項が無視できない場合においては、利得余裕値を精度良く推定することが難しくなる。   In the above basic configuration, the gain margin value is estimated from the response signal Q expressed by Equation 1, but as is clear from Equation 1, the response signal Q includes a response signal (formula 1 except for the first term on the right side of 1). Therefore, when the level of the received signal Y and the sound signal X collected by the microphone 1 is large and the second term and the third term on the right side of Equation 1 cannot be ignored, it is difficult to accurately estimate the gain margin value. .

そこで、本発明の各実施形態においては、式1で表される応答信号Qのサンプル信号に対応する信号(式1の右辺第1項)のみを抽出することにより、その他の信号(式1の右辺第2項及び第3項)を充分に抑圧し、精度良く利得余裕値を推定することができるようにしている。   Therefore, in each embodiment of the present invention, by extracting only the signal corresponding to the sample signal of the response signal Q represented by Expression 1 (the first term on the right side of Expression 1), other signals (Expression 1 The second term and the third term on the right side) are sufficiently suppressed so that the gain margin value can be estimated with high accuracy.

図1は本実施形態における挿入損失量調整手段8の主要な構成を示すブロック図である。なお、本実施形態の拡声通話機の全体構成は上述の基本例と共通であるから図示並びに説明は省略し、共通する部分については同一の符号を付す。   FIG. 1 is a block diagram showing a main configuration of the insertion loss amount adjusting means 8 in the present embodiment. In addition, since the whole structure of the loudspeaker of this embodiment is common with the above-mentioned basic example, illustration and description are abbreviate | omitted and the same code | symbol is attached | subjected about a common part.

図1に示すように本実施形態における挿入損失量調整手段8は、インパルス信号発生器9、加算器10、包絡線検波器11並びに閉ループ利得余裕推定部12に加えて、所定の時間間隔でインパルス信号発生器9から信号経路へ複数回にわたってインパルス信号を送出させるためのタイミング制御を行う同期制御部15と、包絡線検波器11の前段に設けられ同期制御部15からのタイミング情報を受けてインパルス信号に対する応答信号Qを同期加算平均処理する同期加算平均部16とを具備している。   As shown in FIG. 1, in addition to the impulse signal generator 9, the adder 10, the envelope detector 11, and the closed loop gain margin estimation unit 12, the insertion loss amount adjusting means 8 in the present embodiment includes impulses at predetermined time intervals. A synchronization control unit 15 that performs timing control for sending an impulse signal from the signal generator 9 to the signal path a plurality of times, and an impulse that is provided in the preceding stage of the envelope detector 11 and receives timing information from the synchronization control unit 15 And a synchronous addition averaging unit 16 for performing a synchronous addition averaging process on the response signal Q to the signal.

図2は同期制御部15並びに同期加算平均部16を示すブロック図である。同期制御部15は、図3に示すように所定の時間間隔T1,T2,…,TN-1毎に複数回励振されるインパルス信号(サンプル信号)P0,P1,…,PN-1(以下、{P}と表記する。)をインパルス信号発生器9から送出させるとともに、図2に示すようにインパルス信号P0…の送出に同期した同期信号を同期加算平均部16に出力する。 FIG. 2 is a block diagram showing the synchronization control unit 15 and the synchronization addition averaging unit 16. Synchronization control section 15, a predetermined time interval T 1, as shown in FIG. 3, T 2, ..., T N-1 impulse signal to be excited more than once every (sample signal) P 0, P 1, ... , P N-1 (hereinafter referred to as {P}) is sent from the impulse signal generator 9 and a synchronous signal synchronized with the transmission of the impulse signals P 0 ... Is sent to the synchronous addition averaging unit 16 as shown in FIG. Output.

一方、同期加算平均部16は、図2に示すように同期制御部15からの同期信号によってオン・オフされるスイッチ要素16aと、スイッチ要素16aを介して入力されるインパルス信号{P}の応答信号{Q}を所定の時間だけ遅延させる遅延器16b1〜16bN-1と、各遅延器16b1〜16bN-1の出力を加算する加算器16c1〜16cN-1と、全遅延器16b1〜16bN-1の出力信号の和(加算器16cN-1の出力)を同期加算回数Nで除算する除算器16dとを備えて、インパルス信号{P}に同期して加算平均処理を行って応答信号{Q}の移動平均を求めている。而して、音響信号Xや受話信号Yの位相はインパルス信号{P}が送出される時間間隔T1,T2,…,TN-1とは無関係であるため、同期加算回数Nを充分に大きくとることで式1における右辺第2項及び第3項が抑制されるとともに右辺第1項のインパルス信号{P}に対する応答成分のみが強調され、実質的に式1の右辺第1項の上記応答成分のみを抽出することが可能となる。なお、応答信号{Q}の移動平均が求められたら、同期制御部15から出力されるリセット信号により、同期加算平均部16の各遅延器16b1〜16bN-1がリセットされ、同期加算平均部16が新たに移動平均の算出可能な初期状態に復帰する。 On the other hand, as shown in FIG. 2, the synchronous addition averaging unit 16 is turned on / off by the synchronization signal from the synchronization control unit 15, and the response of the impulse signal {P} input through the switch element 16a. a delay unit 16b 1 ~16b N-1 for delaying the signal {Q} predetermined time, the adder 16c 1 ~16c N-1 for adding the outputs of the delay units 16b 1 ~16b N-1, the total delay And a divider 16d that divides the sum of the output signals of the devices 16b 1 to 16b N-1 (the output of the adder 16c N-1 ) by the number N of synchronous additions, and adds and averages in synchronization with the impulse signal {P} Processing is performed to obtain a moving average of the response signal {Q}. Thus, the phase of the acoustic signal X and the reception signal Y is not related to the time intervals T 1 , T 2 ,..., T N-1 at which the impulse signal {P} is transmitted. To suppress the second term and the third term on the right side in Equation 1 and emphasize only the response component for the impulse signal {P} in the first term on the right side. Only the response component can be extracted. When the moving average of the response signal {Q} is obtained, the delay signals 16b 1 to 16b N−1 of the synchronous addition averaging unit 16 are reset by the reset signal output from the synchronous control unit 15, and the synchronous addition average is obtained. The unit 16 newly returns to the initial state where the moving average can be calculated.

そして、同期加算平均部16から出力される応答信号{Q}の移動平均が包絡線検波器11に入力され、同期加算平均処理により得られたインパルス信号{P}に対する閉ループ系の応答信号の包絡線成分が抽出される。さらに、閉ループ利得余裕推定部12では、包絡線検波器11で抽出された包絡線成分の時間軸に対する減衰特性から利得余裕値を推定し、推定された利得余裕値に基づいて所要の挿入損失量を算出する。なお、インパルス信号{P}の時間間隔T1,T2,…,TN-1は、利得余裕値の推定処理に必要な応答時間τに対して充分に大きくする必要がある。また、利得余裕値の推定処理に要する加算器11の出力信号の観測時間T(=T0+T1+…+TN-1+τ)は、この時間T内における閉ループ系の特性の変動が無視できる程度に短くする必要がある。 Then, the moving average of the response signal {Q} output from the synchronous addition averaging unit 16 is input to the envelope detector 11, and the envelope of the closed loop response signal with respect to the impulse signal {P} obtained by the synchronous addition averaging process Line components are extracted. Further, the closed loop gain margin estimation unit 12 estimates a gain margin value from the attenuation characteristic of the envelope component extracted by the envelope detector 11 with respect to the time axis, and a required insertion loss amount based on the estimated gain margin value. Is calculated. Note that the time intervals T 1 , T 2 ,..., T N-1 of the impulse signal {P} need to be sufficiently larger than the response time τ required for the gain margin value estimation process. Further, the observation time T (= T 0 + T 1 +... + T N-1 + τ) of the output signal of the adder 11 required for the process of estimating the gain margin value can ignore the fluctuations in the characteristics of the closed loop system within this time T. It needs to be as short as possible.

上述のように基本構成の挿入損失量調整手段8に同期制御部15と同期加算平均部16とを付加して、インパルス信号{P}に対する応答信号{Q}の移動平均を求め、求めた移動平均を包絡線検波器11に出力するようにしているため、閉ループ中に周囲騒音や音声信号が存在する場合においても、応答信号{Q}に対する周囲騒音の影響を軽減し、利得余裕値を精度良く推定することができる。その結果、上記のような場合においても挿入損失量を適切な値に設定して、閉ループ系を安定化することができる。特に、本手法では、閉ループ中に存在する周囲騒音が広い周波数帯域を持ち、自己相関が少ない場合にその効果が大きい。   As described above, the synchronization control unit 15 and the synchronization addition averaging unit 16 are added to the insertion loss amount adjusting means 8 having the basic configuration, and the moving average of the response signal {Q} with respect to the impulse signal {P} is obtained. Since the average is output to the envelope detector 11, the influence of the ambient noise on the response signal {Q} is reduced and the gain margin value is accurate even when ambient noise and voice signals are present in the closed loop. It can be estimated well. As a result, even in the above case, the closed-loop system can be stabilized by setting the insertion loss amount to an appropriate value. In particular, this method is particularly effective when the ambient noise existing in the closed loop has a wide frequency band and the autocorrelation is small.

ところで、図1に示した構成においては閉ループ系の利得余裕値を推定するためにサンプル信号(インパルス信号又はバースト信号)を系に与えているが、通話中にインパルス信号やバースト信号がスピーカ3から聞こえるために使用者に聴感上の不快感を与えてしまう虞がある。   In the configuration shown in FIG. 1, a sample signal (impulse signal or burst signal) is given to the system in order to estimate the gain margin value of the closed loop system. Since it can be heard, there is a risk of giving the user unpleasant discomfort.

そこで、インパルス信号やバースト信号のように聴感上不快感を与える虞があるサンプル信号を使う代わりに、疑似白色信号を用いて利得余裕値を推定すれば、上述のような聴感上の不快感を与えるのを防止することができる。   Therefore, if the gain margin value is estimated using a pseudo white signal instead of using a sample signal that may cause an unpleasant sensation such as an impulse signal or a burst signal, the above unpleasant sensation on the audibility described above can be obtained. Can be prevented.

図4は本実施形態の挿入損失量調整手段8の他の主要構成を示すブロック図であり、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器24と、音声信号に疑似白色信号を加算する加算器10と、疑似白色信号と当該疑似白色信号に対する応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部25と、相互相関演算部25により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器11と、包絡線検波器11の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部12とを具備している。なお、図1に示したものと共通する部分については同一の符号を付して詳細な説明は省略する。   FIG. 4 is a block diagram showing another main configuration of the insertion loss amount adjusting means 8 of the present embodiment. The pseudo white signal generator 24 inserts a pseudo white signal having a finite period into the signal path, and the audio signal is simulated. An adder 10 for adding a white signal; a cross-correlation calculating unit 25 that calculates a cross-correlation value of a pseudo-white signal and a response signal to the pseudo-white signal and estimates a closed-loop impulse response based on the calculated cross-correlation value; An envelope detector 11 for detecting the envelope of the closed-loop impulse response signal estimated by the cross-correlation calculator 25, and a closed-loop gain margin estimator for estimating the gain margin in the closed loop based on the output signal of the envelope detector 11 12. In addition, the same code | symbol is attached | subjected about the part which is common in what was shown in FIG. 1, and detailed description is abbreviate | omitted.

ここで、疑似白色信号としては、例えばM系列信号を用いる。M系列信号とは、次式のf(x)がmod2の原始多項式であるとき、
f(x)=1+c1x+c22+…cpp
次式により生成されるmod2上の数列atで表される。
Here, for example, an M-sequence signal is used as the pseudo white signal. The M-sequence signal means that f (x) in the following equation is a primitive polynomial of mod 2
f (x) = 1 + c 1 x + c 2 x 2 +... c p x p
Represented by sequence a t on mod2 generated by the following equation.

t=c1t-1+c2t-2+…+cpt-p
また、この周期Tは2p−1である。信号経路へ送出する信号p(t)がM系列信号であり、その周期Tが充分に長ければ、M系列信号p(t)に対して次式が成り立つ。
a t = c 1 a t-1 + c 2 a t-2 +... + c p a tp
The period T is 2 p −1. If the signal p (t) sent to the signal path is an M-sequence signal and its period T is sufficiently long, the following equation holds for the M-sequence signal p (t).

Figure 0004349296
Figure 0004349296

ここで、ΨppはM系列信号p(t)の自己相関値を表すが、M系列信号の自己相関値が時間に依存しないため、次式のように表現することができる。 Here, Ψ pp represents the autocorrelation value of the M-sequence signal p (t). However, since the autocorrelation value of the M-sequence signal does not depend on time, it can be expressed as the following equation.

Ψpp(t、t+τ)=Ψpp(τ)
相互相関演算部25においては、次式によりM系列信号p(t)と加算器10の出力信号(応答信号)q(t)との相互相関値を上記Aで規格化した値を求める。
Ψ pp (t, t + τ) = Ψ pp (τ)
In the cross-correlation calculating unit 25, a value obtained by normalizing the cross-correlation value between the M-sequence signal p (t) and the output signal (response signal) q (t) of the adder 10 with the above-described A is obtained.

Figure 0004349296
Figure 0004349296

ここで、加算器10の出力信号q(t)は次式で表される。 Here, the output signal q (t) of the adder 10 is expressed by the following equation.

Figure 0004349296
Figure 0004349296

但し、hp(σ)、hy(σ)並びにhx(σ)は各々M系列信号p(t)の入力点から加算器10の出力点への閉ループ系のインパルス応答、受話信号Yの入力点から加算器10の出力点への閉ループ系のインパルス応答、送話信号Xの入力点から加算器10の出力点への閉ループ系のインパルス応答をそれぞれ表している。上記式4を式3に代入すると次式が得られる。 However, h p (σ), h y (σ) and h x (σ) are the impulse response of the closed loop system from the input point of the M-sequence signal p (t) to the output point of the adder 10, respectively, The closed loop impulse response from the input point to the output point of the adder 10 and the closed loop impulse response from the input point of the transmission signal X to the output point of the adder 10 are shown. Substituting Equation 4 into Equation 3 yields:

Figure 0004349296
Figure 0004349296

上記式5は次式のように変形できる。 The above equation 5 can be transformed as the following equation.

Figure 0004349296
Figure 0004349296

但し、Ψpy及びΨpxはそれぞれM系列信号p(t)と受話信号Yとの相互相関値、M系列信号p(t)と送話信号Xとの相互相関値を表し、各々次式で与えられる。 Where ψ py and ψ px represent the cross-correlation value between the M-sequence signal p (t) and the received signal Y, and the cross-correlation value between the M-sequence signal p (t) and the transmitted signal X, respectively. Given.

Figure 0004349296
Figure 0004349296

Figure 0004349296
Figure 0004349296

ここで、M系列信号p(t)と受話信号Y、並びにM系列信号p(t)と送話信号Xとは互いに因果関係がないため、それぞれの相互相関値Ψpy、Ψpxは時間tに依存して変化する。また、M系列信号p(t)の周期Tが充分に長いときには上記相互相関値Ψpy、Ψpxはゼロとなる。よって、この場合には上記式6の右辺第2項及び第3項がゼロとなり、次式のように変形できる。 Here, since the M-sequence signal p (t) and the received signal Y, and the M-sequence signal p (t) and the transmitted signal X are not causal to each other, the respective cross-correlation values ψ py and ψ px are the time t. Varies depending on When the period T of the M-sequence signal p (t) is sufficiently long, the cross-correlation values ψ py and ψ px are zero. Therefore, in this case, the second term and the third term on the right side of Equation 6 are zero and can be transformed as the following equation.

Figure 0004349296
Figure 0004349296

従って、Ψpp(t、t+τ)=Ψpp(τ)で与えられるM系列信号(疑似白色信号)p(t)と応答信号q(t)との相互相関値を上記Aで規格化した値を算出することにより、M系列信号p(t)から応答信号q(t)への系のインパルス応答の遅延時間τにおける係数を求めることができる。但し、上記式9が成り立つのは、式7及び式8で与えられる相互相関値Ψpy,ΨpxがゼロとみなせるほどM系列信号p(t)の周期Tが充分に長い場合であり、それ以外の場合においては、上記式6の右辺第2項及び第3項は外乱成分となり、インパルス応答の推定精度を劣化させる要因となる。 Therefore, the value obtained by normalizing the cross-correlation value between the M-sequence signal (pseudo white signal) p (t) given by Ψ pp (t, t + τ) = Ψ pp (τ) and the response signal q (t) by A. Is obtained, the coefficient in the delay time τ of the impulse response of the system from the M-sequence signal p (t) to the response signal q (t) can be obtained. However, the above equation 9 holds when the period T of the M-sequence signal p (t) is sufficiently long that the cross-correlation values ψ py and ψ px given by the equations 7 and 8 can be regarded as zero. In the other cases, the second term and the third term on the right-hand side of Equation 6 above become disturbance components, which are factors that degrade the estimation accuracy of the impulse response.

ところで、相互相関演算部25をデジタル回路により実現する場合には、上記Ψpp(t、t+τ)=Ψpp(τ)を上記Aで規格化した相互相関値は次式で表される By the way, when the cross-correlation calculation unit 25 is realized by a digital circuit, the cross-correlation value obtained by normalizing the above Ψ pp (t, t + τ) = Ψ pp (τ) with the above A is expressed by the following equation.

Figure 0004349296
Figure 0004349296

但し、Iは離散時間系におけるM系列信号p(t)の周期を表し、p(i)、q(i+j)は各々サンプル時刻i,i+jにおけるM系列信号p(t)及び応答信号q(t)のデータを表す。また、上記式7により得られる値はM系列信号p(t)から応答信号q(t)への系のインパルス応答の1係数hp(j)であり、閉ループ系の利得余裕を推定するために必要なインパルス応答観測時間をサンプル時間Jとすると、hp(0),h(1),h(2),…,h(J-1)の合計J個の係数を求める必要がある。よって、M系列信号(疑似白色信号)p(t)から応答信号q(t)への系のインパルス応答を推定するために、I×J回の積和演算を必要とする。 Here, I represents the period of the M-sequence signal p (t) in the discrete time system, and p (i) and q (i + j) are the M-sequence signal p (t) and the response signal q at the sample times i and i + j, respectively. Represents the data of (t). Further, the value obtained by the above equation 7 is one coefficient h p (j) of the impulse response of the system from the M-sequence signal p (t) to the response signal q (t), and is used to estimate the gain margin of the closed loop system. If the impulse response observation time required for the above is sample time J, it is necessary to obtain a total of J coefficients of h p (0), h (1), h (2),..., H (J−1). Therefore, in order to estimate the impulse response of the system from the M-sequence signal (pseudo white signal) p (t) to the response signal q (t), I × J product-sum operations are required.

上述のような演算処理により求められる疑似白色信号p(t)から応答信号q(t)への系のインパルス応答(応答信号Q)が包絡線検波器11に入力され、閉ループ利得余裕推定部12において包絡線検波器11で抽出された包絡線成分の時間軸に対する減衰特性から利得余裕値を推定し、推定された利得余裕値に基づいて所要の挿入損失量を算出する。   The impulse response (response signal Q) of the system from the pseudo white signal p (t) to the response signal q (t) obtained by the arithmetic processing as described above is input to the envelope detector 11 and the closed loop gain margin estimation unit 12 The gain margin value is estimated from the attenuation characteristic of the envelope component extracted by the envelope detector 11 with respect to the time axis, and the required insertion loss amount is calculated based on the estimated gain margin value.

上述のように、基本構成の挿入損失量調整手段8に対してインパルス信号発生器9の代わりに疑似白色信号発生器24を付加するとともに、疑似白色信号と当該疑似白色信号に対する応答信号の相互相関値を求めて相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部25を付加して、推定したインパルス応答(応答信号)から包絡線検波器11並びに閉ループ利得余裕推定部12により利得余裕値を推定するため、使用者に聴感上の不快感を与えることなく利得余裕値を精度よく推定でき、挿入損失量を適切な値に設定して、閉ループ系を安定化することができる。   As described above, the pseudo white signal generator 24 is added in place of the impulse signal generator 9 to the insertion loss amount adjusting means 8 of the basic configuration, and the cross correlation between the pseudo white signal and the response signal to the pseudo white signal is added. A cross-correlation calculating unit 25 that obtains a value and estimates a closed-loop impulse response based on the cross-correlation value is added, and the envelope detector 11 and the closed-loop gain margin estimating unit 12 gain from the estimated impulse response (response signal). Since the margin value is estimated, the gain margin value can be accurately estimated without giving the user unpleasant discomfort, and the closed loop system can be stabilized by setting the insertion loss amount to an appropriate value.

図5は本実施形態における挿入損失量調整手段8のさらに他の主要構成を示すブロック図であり、図4に示した構成に対して図1に示した構成における同期制御部15並びに同期加算平均部16を設けた点に特徴がある。   FIG. 5 is a block diagram showing still another main configuration of the insertion loss amount adjusting means 8 in the present embodiment. The synchronization control unit 15 and the synchronous addition average in the configuration shown in FIG. 1 with respect to the configuration shown in FIG. The feature is that the portion 16 is provided.

本構成では、信号経路に送出する疑似白色信号の周期が充分に長くなく、上記式6の右辺第2項及び第3項がゼロとはみなせずに外乱要素となる場合に、同期加算平均処理によりこれらの影響を低減し、閉ループ系のインパルス応答の推定精度を向上させるものである。いま、M系列信号p(t)の周期をT1、同期加算回数をNとすると、同期加算平均部16の出力値は次式で表される。 In this configuration, when the period of the pseudo white signal transmitted to the signal path is not sufficiently long and the second term and the third term on the right side of Equation 6 are not regarded as zero and become a disturbance element, a synchronous addition averaging process is performed. Therefore, these effects are reduced and the estimation accuracy of the impulse response of the closed loop system is improved. Now, assuming that the period of the M-sequence signal p (t) is T 1 and the number of synchronous additions is N, the output value of the synchronous addition averaging unit 16 is expressed by the following equation.

Figure 0004349296
Figure 0004349296

上記式11に式6及び式9を代入して整理すると次式が得られる。 Substituting Equations 6 and 9 into Equation 11 above and rearranging results in the following equation.

Figure 0004349296
Figure 0004349296

ここで、相互相関値Ψpy,Ψpxは時間tに関して不規則に変動するため、上記式12における右辺第2項及び第3項が同期加算回数Nの値を充分に大きくすることにより抑圧することができる。 Here, since the cross-correlation values Ψ py and Ψ px fluctuate irregularly with respect to time t, the second term and the third term on the right side in the above equation 12 are suppressed by sufficiently increasing the value of the number of times of synchronous addition N. be able to.

本構成によれば、信号経路に送出するM系列信号p(t)の周期Tを図4に示した構成に比較して短くすることができるため、挿入損失量調整手段8をデジタル回路で実現する場合にM系列信号p(t)の発生及び上記式10の演算に必要となる記憶装置の容量や積和回数を低減することができるという利点がある。なお、同期加算平均部16より出力される閉ループ系のインパルス応答(応答信号)は包絡線検波器11に入力され、図1に示した構成と同様の演算処理によって所要の挿入損失量が算出される。   According to this configuration, the period T of the M-sequence signal p (t) transmitted to the signal path can be shortened as compared with the configuration shown in FIG. 4, so that the insertion loss amount adjusting means 8 is realized by a digital circuit. In this case, there is an advantage that the capacity of the storage device and the number of product sums required for the generation of the M-sequence signal p (t) and the calculation of Equation 10 can be reduced. The closed loop impulse response (response signal) output from the synchronous addition averaging unit 16 is input to the envelope detector 11, and the required insertion loss amount is calculated by the same arithmetic processing as the configuration shown in FIG. The

(実施形態1)
図6は本発明の実施形態1における挿入損失量調整手段8を示すブロック図であり、図1に示した構成に対して収束判定部17を設けた点に特徴がある。
(Embodiment 1)
FIG. 6 is a block diagram showing the insertion loss amount adjusting means 8 according to the first embodiment of the present invention, which is characterized in that a convergence determining unit 17 is provided in the configuration shown in FIG.

利得余裕値の推定処理を開始した後、M番目のインパルス信号PM-1を閉ループに送出した後に同期加算平均部15から出力される信号QM’(t)は次式で表される。但し、0≦t≦τであり、t=0は閉ループ系にM番目のインパルス信号PM-1を送出した瞬間を表す。 After the gain margin value estimation process is started, the signal Q M '(t) output from the synchronous addition averaging unit 15 after the Mth impulse signal P M-1 is sent to the closed loop is expressed by the following equation. However, 0 ≦ t ≦ τ, and t = 0 represents the moment when the Mth impulse signal P M-1 is sent to the closed loop system.

Figure 0004349296
Figure 0004349296

さらに、M+1番目のインパルス信号PMを送出した後に同期加算平均部16より出力される信号QM+1’(t)は次式で表される。 Further, a signal Q M + 1 ′ (t) output from the synchronous addition averaging unit 16 after sending the M + 1-th impulse signal P M is expressed by the following equation.

Figure 0004349296
Figure 0004349296

ここで、収束判定部17においては、同期加算平均部16の出力信号Q’(t)を常時監視し、M+1回目の同期加算平均処理により得られる信号QM+1’(t)と前回の同期加算平均処理により得られた信号QM’(t)との差が一定の範囲内に収束したか否かを判定する。すなわち、収束判定の規範は次式で与えられる。 Here, the convergence determination unit 17 constantly monitors the output signal Q ′ (t) of the synchronous addition averaging unit 16, and the signal Q M + 1 ′ (t) obtained by the M + 1th synchronous addition averaging process and the previous time. It is determined whether the difference from the signal Q M ′ (t) obtained by the synchronous addition averaging process has converged within a certain range. That is, the convergence criterion is given by the following equation.

Figure 0004349296
Figure 0004349296

上式は式14を用いて次式のように変形できる。   The above equation can be transformed into the following equation using equation 14.

Figure 0004349296
Figure 0004349296

上式より、式15の収束条件を満たすためには、αが十分に小さく(すなわち、同期加算回数Mが十分に大きく)、かつ、同期加算平均部16の出力信号Q’(t)が応答信号Q(t)の期待値E〔Q(t)〕に収束する必要があることが分かる。   From the above equation, in order to satisfy the convergence condition of Equation 15, α is sufficiently small (that is, the number of times of synchronous addition M is sufficiently large), and the output signal Q ′ (t) of the synchronous addition averaging unit 16 responds. It can be seen that it is necessary to converge to the expected value E [Q (t)] of the signal Q (t).

而して、収束判定部17は、同期加算平均部16の出力信号Q’(t)の収束が確認されるまで同期加算処理を更新し、収束が確認された時点で同期制御部15を介して同期加算平均部16における同期加算処理を中断させるとともにインパルス信号発生器9からのインパルス信号{P}の送出を停止させる。なお、同期加算平均部16の出力信号Q’(t)は包絡線検波器11に入力され、以下、図1に示した構成と同様の処理により所要の挿入損失量が算出される。   Thus, the convergence determination unit 17 updates the synchronization addition process until the convergence of the output signal Q ′ (t) of the synchronization addition averaging unit 16 is confirmed, and when the convergence is confirmed, the convergence determination unit 17 passes the synchronization control unit 15. Thus, the synchronous addition processing in the synchronous addition averaging unit 16 is interrupted and the transmission of the impulse signal {P} from the impulse signal generator 9 is stopped. The output signal Q '(t) of the synchronous addition averaging unit 16 is input to the envelope detector 11, and the required insertion loss amount is calculated by the same processing as in the configuration shown in FIG.

上述のように本実施形態では、収束判定部17の判定結果に応じて同期制御部15が同期加算平均部16における同期加算処理の継続/中断を決定しているので、同期加算平均部16の出力信号Q’(t)が一定値に収束するまでインパルス信号{P}の送出と同期加算処理とを継続することによって、閉ループ中に周囲騒音が存在する場合においても、周囲騒音が存在しないときと同程度の精度で利得余裕値を推定することが可能となる。なお、図5に示した構成に対して収束判定部17を設けても同様の作用効果を奏することができる。   As described above, in the present embodiment, the synchronization control unit 15 determines continuation / interruption of the synchronization addition process in the synchronization addition averaging unit 16 according to the determination result of the convergence determination unit 17. When the ambient noise does not exist even when ambient noise exists in the closed loop by continuing the transmission of the impulse signal {P} and the synchronous addition process until the output signal Q ′ (t) converges to a constant value. It is possible to estimate the gain margin value with the same degree of accuracy. In addition, even if the convergence determination part 17 is provided with respect to the structure shown in FIG. 5, there can exist the same effect.

(実施形態2)
図7は本発明の実施形態2における挿入損失量調整手段8を示すブロック図であり、実施形態1に対してタイマ部18を設けた点に特徴がある。
(Embodiment 2)
FIG. 7 is a block diagram showing the insertion loss amount adjusting means 8 in the second embodiment of the present invention, which is characterized in that a timer unit 18 is provided in the first embodiment.

タイマ部18は、インパルス信号発生器9から最初のインパルス信号P0が送出されると同時に同期制御部15によってリセットされ、それ以後の経過時間を測定する。この経過時間を示す信号がタイマ部18から同期制御部15に随時入力されており、同期制御部15では、経過時間が所定時間を越えても収束判定部17にて同期加算平均部16の出力信号Q’(t)の収束が確認されなければ、同期加算平均部16にリセット信号を出力して利得余裕推定処理を中断するとともに、インパルス信号発生器9からのインパルス信号{P}の送出を停止させる。而して、インパルス信号{P}の送出中に閉ループ系が変動し、長時間にわたって同期加算処理を行ってもその出力信号Q’(t)が収束しないような場合において、同期加算平均処理を一度リセットすることにより、閉ループ系が変動した後の利得余裕値を速やかに推定することができる。 The timer unit 18 is reset by the synchronization control unit 15 at the same time as the first impulse signal P 0 is sent from the impulse signal generator 9 and measures the elapsed time thereafter. A signal indicating the elapsed time is input from the timer unit 18 to the synchronization control unit 15 at any time. In the synchronization control unit 15, even if the elapsed time exceeds a predetermined time, the convergence determination unit 17 outputs the synchronization addition averaging unit 16. If the convergence of the signal Q ′ (t) is not confirmed, a reset signal is output to the synchronous addition averaging unit 16 to interrupt the gain margin estimation process, and the impulse signal {P} is transmitted from the impulse signal generator 9. Stop. Thus, in the case where the closed loop system fluctuates during transmission of the impulse signal {P} and the output signal Q ′ (t) does not converge even if the synchronous addition processing is performed for a long time, the synchronous addition averaging processing is performed. By resetting once, it is possible to quickly estimate the gain margin value after the closed-loop system fluctuates.

上述のように本実施形態によれば、利得余裕値の推定処理中に閉ループ系の特性が変化し、タイマ部18により測定される経過時間が所定時間を越えても同期加算平均部16の出力信号Q’(t)が収束しない場合には、一度同期加算処理を中断させてリセットすることで変化後の閉ループ系の利得余裕値の推定処理を速やかに開始することができる。なお、図5に示した構成に対して収束判定部17とともにタイマ部18を設けても同様の作用効果を奏することができる。   As described above, according to the present embodiment, the characteristics of the closed loop system change during the gain margin value estimation process, and the output of the synchronous addition averaging unit 16 even if the elapsed time measured by the timer unit 18 exceeds a predetermined time. When the signal Q ′ (t) does not converge, the process of estimating the gain margin value of the closed loop system after the change can be started quickly by interrupting and resetting the synchronous addition process once. In addition, even if the timer unit 18 is provided together with the convergence determination unit 17 in the configuration shown in FIG.

(実施形態3)
図8は本発明の実施形態3における挿入損失量調整手段8を示すブロック図であり、図1に示した構成に対して自己相関関数算出部19を設けた点に特徴がある。
(Embodiment 3)
FIG. 8 is a block diagram showing the insertion loss amount adjusting means 8 according to the third embodiment of the present invention, which is characterized in that an autocorrelation function calculation unit 19 is provided in the configuration shown in FIG.

自己相関関数算出部19は、インパルス信号P0…の送出以前に閉ループ中に存在する信号の自己相関関数を算出するものである。ここで、インパルス信号P0…の送出以前に閉ループ中に存在する信号をr(t)とし、その自己相関関数をR(τ)とすると、この自己相関関数R(τ)は次式で表される。 The autocorrelation function calculation unit 19 calculates an autocorrelation function of a signal existing in the closed loop before sending the impulse signals P 0 . Here, if the signal existing in the closed loop before sending the impulse signals P 0 ... Is r (t) and the autocorrelation function is R (τ), the autocorrelation function R (τ) is expressed by the following equation. Is done.

Figure 0004349296
Figure 0004349296

ここで、インパルス信号{P}に対する応答信号{Q}は、インパルス信号P0…に対する応答成分と元々閉ループ中に存在している信号r(t)(雑音成分)との和で表される。従って、時間シフト量τと、所定値以上の雑音成分の抑圧量を得るために必要な同期加算回数Nとの関係を事前に調べておき、同期制御部15にて、その関係と自己相関関数算出部19により得られる自己相関関数R(τ)とを照合することにより、同期加算処理に要する時間T(=T0+T1+…+TN-1+τ)を最小化するためのインパルス信号{P}の送出間隔Tn(n=0,1,…,N−1)及び同期加算回数Nを求めることができる。その結果、利得余裕値の推定に要する時間を短縮することができる。 Here, the response signal {Q} to the impulse signal {P} is represented by the sum of the response component to the impulse signal P 0 ... And the signal r (t) (noise component) originally present in the closed loop. Accordingly, the relationship between the time shift amount τ and the number of times of synchronization addition N necessary for obtaining a noise component suppression amount equal to or greater than a predetermined value is examined in advance, and the synchronization control unit 15 determines the relationship and the autocorrelation function. By comparing with the autocorrelation function R (τ) obtained by the calculation unit 19, an impulse signal {for minimizing the time T (= T 0 + T 1 +... + T N-1 + τ) required for the synchronous addition process { The transmission interval T n (n = 0, 1,..., N−1) and the number N of synchronous additions can be obtained. As a result, the time required to estimate the gain margin value can be shortened.

上述のように本実施形態では、閉ループ中に存在する信号r(t)の自己相関関数R(τ)を算出する自己相関関数算出部19を挿入損失量調整手段8に設け、算出された自己相関関数R(τ)に基づいて同期制御部15がインパルス信号発生器9からのインパルス信号{P}の送出間隔Tnと同期加算平均部16における同期加算回数Nを調整するようにしたので、事前に算出された閉ループ中に存在する信号r(t)の自己相関関数R(τ)より、同期加算処理に要する時間Tを最小化するようにインパルス信号{P}の送出間隔Tn及び同期加算回数Nを調整することで利得余裕値の推定処理を高速化することができる。なお、図5に示した構成に対して自己相関関数算出部19を設けても同様の作用効果を奏することができる。 As described above, in the present embodiment, the auto-correlation function calculating unit 19 that calculates the auto-correlation function R (τ) of the signal r (t) existing in the closed loop is provided in the insertion loss amount adjusting unit 8, and the calculated self Since the synchronization control unit 15 adjusts the transmission interval T n of the impulse signal {P} from the impulse signal generator 9 and the number N of synchronization additions in the synchronization addition averaging unit 16 based on the correlation function R (τ). From the autocorrelation function R (τ) of the signal r (t) existing in the closed loop calculated in advance, the transmission interval T n and the synchronization of the impulse signal {P} so as to minimize the time T required for the synchronous addition process The gain margin value estimation process can be speeded up by adjusting the number N of additions. In addition, even if the autocorrelation function calculation unit 19 is provided in the configuration shown in FIG.

(実施形態4)
図9は本発明の実施形態4における挿入損失量調整手段8を示すブロック図であり、図1に示した構成に対して騒音レベル推定部20とインパルス信号送出レベル調整部21とを設けた点に特徴がある。
(Embodiment 4)
FIG. 9 is a block diagram showing the insertion loss amount adjusting means 8 according to the fourth embodiment of the present invention, in which a noise level estimation unit 20 and an impulse signal transmission level adjustment unit 21 are provided in the configuration shown in FIG. There is a feature.

騒音レベル推定部20は、インパルス信号{P}の閉ループ系への送出以前に閉ループ中に存在する周囲騒音のレベルを推定するものであり、インパルス信号送出レベル調整部21は、推定された周囲騒音のレベルに応じてインパルス信号発生器9から出力されるインパルス信号P0…のレベルを調整するものである。すなわち、本発明に係る利得余裕値の推定処理においては、閉ループ中の周囲騒音レベルに対するインパルス信号送出レベルの比(この比をここではS/N比と定義する)が利得余裕値の推定精度に影響を及ぼすことから、閉ループにインパルス信号P0…が送出される以前の周囲騒音レベルを騒音レベル推定部20で推定し、推定された周囲騒音レベルに対して上記S/N比が所定値以上となるようにインパルス信号送出レベル調整部21にてインパルス信号P0…の送出レベルを調整することで、利得余裕値の推定精度を向上させることができる。 The noise level estimation unit 20 estimates the level of ambient noise existing in the closed loop before the impulse signal {P} is sent to the closed loop system, and the impulse signal transmission level adjustment unit 21 estimates the ambient noise. The level of the impulse signals P 0 ... Output from the impulse signal generator 9 is adjusted in accordance with the level of. That is, in the gain margin value estimation process according to the present invention, the ratio of the impulse signal transmission level to the ambient noise level in the closed loop (this ratio is defined here as the S / N ratio) is the gain margin value estimation accuracy. Therefore, the noise level estimation unit 20 estimates the ambient noise level before the impulse signal P 0 ... Is sent to the closed loop, and the S / N ratio is greater than or equal to a predetermined value with respect to the estimated ambient noise level. By adjusting the transmission level of the impulse signals P 0 ... By the impulse signal transmission level adjusting unit 21 so that the following can be obtained, it is possible to improve the estimation accuracy of the gain margin value.

上述のように本実施形態では、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部20と、騒音レベル推定部20による推定結果に基づいてサンプル信号(インパルス信号)の送出レベルを調整するインパルス信号送出レベル調整部21とを挿入損失量調整手段8に設けたので、閉ループ系に送出するインパルス信号のレベルを周囲騒音レベルに対して充分な大きさに設定することにより、観測点において高いS/N比でインパルス信号{P}に対する応答信号{Q}を得ることができ、利得余裕値の推定精度を向上させることができる。なお、図5に示した構成に対して騒音レベル推定部20並びに疑似白色信号の送出レベルを調整する手段を設けても同様の作用効果を奏することができる。   As described above, in this embodiment, the noise level estimation unit 20 that estimates the level of ambient noise existing in the closed loop, and the transmission level of the sample signal (impulse signal) are adjusted based on the estimation result by the noise level estimation unit 20. Since the insertion loss amount adjusting unit 8 is provided with the impulse signal transmission level adjusting unit 21 that performs the above operation, the level of the impulse signal transmitted to the closed loop system is set to a sufficient level with respect to the ambient noise level. The response signal {Q} to the impulse signal {P} can be obtained with a high S / N ratio, and the estimation accuracy of the gain margin value can be improved. It is to be noted that the same effect can be obtained by providing the noise level estimation unit 20 and means for adjusting the transmission level of the pseudo white signal in the configuration shown in FIG.

(実施形態5)
図10は本発明の実施形態5における挿入損失量調整手段8を示すブロック図であり、図1に示した構成に対して騒音レベル推定部20と同期加算回数調整部22とを設けた点に特徴がある。
(Embodiment 5)
FIG. 10 is a block diagram showing the insertion loss amount adjusting means 8 in Embodiment 5 of the present invention, in that a noise level estimating unit 20 and a synchronous addition number adjusting unit 22 are provided in the configuration shown in FIG. There are features.

騒音レベル推定部20は実施形態4と同様に、インパルス信号{P}の閉ループ系への送出以前に閉ループ中に存在する周囲騒音のレベルを推定するものであり、同期加算回数調整部22は、推定された周囲騒音のレベルに応じて同期加算平均部16における同期加算回数Nを調整するものである。一般に、周囲騒音のレベルが大きい場合に周囲騒音を抑圧するためには同期加算回数Nを大きな値に設定する必要があり、逆に、周囲騒音のレベルが小さい場合には少ない同期加算回数Nでも周囲騒音を抑圧することができる。   Similarly to the fourth embodiment, the noise level estimation unit 20 estimates the level of ambient noise existing in the closed loop before sending the impulse signal {P} to the closed loop system. The number of synchronous additions N in the synchronous addition averaging unit 16 is adjusted according to the estimated level of ambient noise. Generally, in order to suppress the ambient noise when the ambient noise level is high, it is necessary to set the number of synchronization additions N to a large value. Conversely, when the ambient noise level is low, even if the number of synchronization additions N is small. Ambient noise can be suppressed.

而して、このような周囲騒音レベルと同期加算回数との統計的性質を予め定量的に調べておき、騒音レベル推定部20で推定される周囲騒音レベルに対して同期加算回数調整部22により必要最小限の同期加算回数Nを設定することで、同期加算平均に伴う遅延時間を最小化することができる。   Thus, the statistical properties of the ambient noise level and the number of synchronous additions are quantitatively examined in advance, and the synchronous addition number adjusting unit 22 is used for the ambient noise level estimated by the noise level estimating unit 20. By setting the necessary minimum number N of synchronous additions, the delay time associated with the average of synchronous additions can be minimized.

上述のように本実施形態では、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部20と、騒音レベル推定部20による推定結果に基づいて同期加算平均部16における同期加算回数Nを調整する同期加算回数調整部22とを挿入損失量調整手段8に設けたので、閉ループ系に存在する周囲騒音のレベルに応じて同期加算回数Nを必要最小限に設定することができるため、同期加算処理に要する計算時間を最小化することができる。また、1回の利得余裕値の推定処理に要するサンプル信号(インパルス信号)の送出回数を最小化することにより、インパルス信号が受話信号に重畳して聞こえることによる聴覚上の不快感、違和感を軽減することができる。なお、図5に示した構成に対して騒音レベル推定部20並びに同期加算回数調整部22を設けても同様の作用効果を奏することができる。   As described above, in the present embodiment, the noise level estimation unit 20 that estimates the level of ambient noise existing in the closed loop, and the synchronization addition number N in the synchronization addition averaging unit 16 based on the estimation result by the noise level estimation unit 20 are calculated. Since the synchronization loss number adjusting unit 22 to be adjusted is provided in the insertion loss amount adjusting means 8, the synchronization addition number N can be set to the minimum necessary according to the level of ambient noise existing in the closed loop system. The calculation time required for the addition process can be minimized. In addition, by minimizing the number of times a sample signal (impulse signal) is sent for one gain margin estimation process, it is possible to reduce auditory discomfort and discomfort due to the impulse signal being heard superimposed on the received signal. can do. In addition, even if the noise level estimation unit 20 and the synchronous addition number adjustment unit 22 are provided in the configuration shown in FIG.

(実施形態6)
図11は本発明の実施形態6における挿入損失量調整手段8を示すブロック図であり、図1に示した構成に対して騒音レベル推定部20、インパルス信号送出レベル調整部21、同期加算回数調整部22並びに周囲騒音抽出部23を設けた点に特徴がある。但し、インパルス信号送出レベル調整部21及び同期加算回数調整部22については、少なくとも何れか一方を設ければよい。また、騒音レベル推定部20、インパルス信号送出レベル調整部21及び同期加算回数調整部22は実施形態4及び実施形態5と共通であるから説明を省略する。
(Embodiment 6)
FIG. 11 is a block diagram showing the insertion loss amount adjusting means 8 according to the sixth embodiment of the present invention. The noise level estimating unit 20, the impulse signal transmission level adjusting unit 21, and the number of times of synchronous addition are adjusted with respect to the configuration shown in FIG. The feature is that the unit 22 and the ambient noise extraction unit 23 are provided. However, at least one of the impulse signal transmission level adjusting unit 21 and the synchronous addition number adjusting unit 22 may be provided. Further, since the noise level estimation unit 20, the impulse signal transmission level adjustment unit 21, and the synchronous addition number adjustment unit 22 are common to the fourth and fifth embodiments, the description thereof is omitted.

式1からも明らかなように、閉ループへのインパルス信号{P}の送出中において観測される応答信号qには、インパルス信号{P}に対する応答成分と周囲騒音成分(および送受話音声成分)が混在している。そこで、周囲騒音抽出部23によってこれら複数の信号成分が重畳された信号から周囲騒音成分のみを抽出し、抽出した成分から周囲騒音レベルを推定することで、インパルス信号{P}の送出後の利得余裕値推定処理中においても、周囲騒音のレベルに応じて適応的にインパルス信号送出レベルや、同期加算回数Nを調整することができるのである。   As is clear from Equation 1, the response signal q observed during transmission of the impulse signal {P} to the closed loop includes a response component for the impulse signal {P} and an ambient noise component (and transmission / reception voice component). It is mixed. Therefore, the ambient noise extraction unit 23 extracts only the ambient noise component from the signal on which the plurality of signal components are superimposed, and estimates the ambient noise level from the extracted component, whereby the gain after transmission of the impulse signal {P} is obtained. Even during the margin value estimation process, the impulse signal transmission level and the number of synchronous additions N can be adaptively adjusted according to the level of ambient noise.

周囲騒音抽出部23は、図12に示すように同期制御部15からインパルス信号Pk(k=0,1,…,N-1)の送出タイミングに同期した同期信号が入力されるとリセットされるタイマ回路23aと、タイマ回路23aにより計測される経過時間tk(各インパルス信号Pkが閉ループ系に送出されてからの経過時間)とインパルス信号Pkに対する閉ループ系の応答時間τとの大小判定を行う判定部23bと、閉ループ系の応答信号Qと判定部23bの出力信号とを乗算するための乗算器23cとを備えて構成される。ここで、乗算器23cの出力信号は、騒音レベル推定部20に入力される。判定部23bは、タイマ回路23aで計測される経過時間tkが応答時間τよりも大きい(tk>τ)と判定した場合には1を出力し、経過時間tkが応答時間τよりも小さい(tk<τ)と判定した場合には0を出力するものである。 The ambient noise extraction unit 23 is reset when a synchronization signal synchronized with the transmission timing of the impulse signal P k (k = 0, 1,..., N−1) is input from the synchronization control unit 15 as shown in FIG. Between the timer circuit 23a and the elapsed time t k (elapsed time since each impulse signal P k is sent to the closed loop system) measured by the timer circuit 23a and the response time τ of the closed loop system with respect to the impulse signal P k A determination unit 23b that performs the determination, and a multiplier 23c that multiplies the closed-loop response signal Q by the output signal of the determination unit 23b. Here, the output signal of the multiplier 23 c is input to the noise level estimation unit 20. The determination unit 23b outputs 1 when it is determined that the elapsed time t k measured by the timer circuit 23a is larger than the response time τ (t k > τ), and the elapsed time t k is longer than the response time τ. If it is determined to be small (t k <τ), 0 is output.

すなわち、閉ループ系にインパルス信号Pkを送出してからの経過時間tkが応答時間τよりも大きく且つ送出間隔Tk+1よりも小さい時間帯(τ<tk<Tk+1)においては、応答信号Qに含まれるインパルス信号Pkに対する応答成分(式1における右辺第1項)は無視でき、τ<tkを満たす時間帯における加算器10の出力信号(応答信号)Qでは周囲騒音に対する成分(及び送受話音声信号に対する成分)が支配的となるため、上述のように経過時間tkが応答時間τよりも大きい(tk>τ)場合に応答信号Qをそのまま騒音レベル推定部20に出力し、経過時間tkが応答時間τ以下(tk<τ)の場合に騒音レベル推定部20に信号を出力しないことによって、周囲騒音抽出部23にて閉ループへのインパルス信号{P}の送出中に観測される応答信号qより周囲騒音成分を抽出することが可能となる。 That is, in a time period (τ <t k <T k + 1 ) in which the elapsed time t k after sending the impulse signal P k to the closed loop system is larger than the response time τ and smaller than the sending interval T k + 1 . Can ignore the response component (the first term on the right side in Equation 1) with respect to the impulse signal P k included in the response signal Q, and the output signal (response signal) Q of the adder 10 in the time zone satisfying τ <t k Since the noise component (and the component for the transmission / reception voice signal) becomes dominant, as described above, when the elapsed time t k is longer than the response time τ (t k > τ), the response signal Q is directly estimated as the noise level. When the elapsed time t k is equal to or shorter than the response time τ (t k <τ), the signal is not output to the noise level estimation unit 20 so that the ambient noise extraction unit 23 outputs an impulse signal { Watch while sending P} Is the it is possible to extract the ambient noise component than the response signal q.

そして、その抽出された周囲騒音成分から騒音レベル推定部20で周囲騒音のレベルを推定し、その推定値に基づき、インパルス信号送出レベル調整部21及び同期加算回数調整部22の少なくとも何れか一方でインパルス信号の送出レベル並びに同期加算回数Nの調整(設定)を行う。なお、周囲騒音抽出部23にて周囲騒音を抽出していない時間帯(tk≦τを満たす時間帯)においては、騒音レベル推定部20での騒音レベルの推定処理を停止するのが望ましい。 Then, the noise level estimation unit 20 estimates the ambient noise level from the extracted ambient noise components, and based on the estimated value, at least one of the impulse signal transmission level adjustment unit 21 and the synchronous addition number adjustment unit 22 is used. Adjustment (setting) of the transmission level of the impulse signal and the number N of synchronous additions is performed. It should be noted that it is desirable to stop the noise level estimation processing in the noise level estimation unit 20 in a time zone in which the ambient noise extraction unit 23 does not extract ambient noise (a time zone satisfying t k ≦ τ).

上述のように本実施形態によれば、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部23を挿入損失量調整手段8に設けたので、利得余裕値の推定処理中に閉ループ中の周囲騒音レベルが変化するような場合においても、これに適応してインパルス信号{P}の送出レベル及び同期加算回数Nの少なくとも何れか一方を調整し、周囲騒音のレベルが変動するような場合においても、これに追従して精度良く利得余裕値を推定することができる。なお、図5に示した構成に対して騒音レベル推定部20、インパルス信号送出レベル調整部21、同期加算回数調整部22並びに周囲騒音抽出部23を設けても同様の作用効果を奏することができる。   As described above, according to the present embodiment, since the ambient noise extraction unit 23 that extracts the ambient noise signal from the signal transmitted during the closed loop is provided in the insertion loss amount adjusting unit 8, the gain margin value estimation process is performed. Even when the ambient noise level in the closed loop changes, at least one of the transmission level of the impulse signal {P} and the number of times of synchronous addition N is adjusted to adapt to this, so that the ambient noise level varies. Even in this case, the gain margin value can be accurately estimated following this. It is to be noted that the same effect can be obtained even if the noise level estimation unit 20, the impulse signal transmission level adjustment unit 21, the synchronous addition number adjustment unit 22, and the ambient noise extraction unit 23 are provided in the configuration shown in FIG. .

ころで、上述の各実施形態では閉ループ系にサンプル信号(インパルス信号又はバースト信号)を送出したときの応答信号の包絡線の傾きを観測することで閉ループの利得余裕値を推定しているが、推定可能な利得余裕値には上限が存在する。つまり、閉ループ系の利得余裕値が所定の値以上(インパルス信号に対する応答信号の包絡線の傾きが所定値以上に大きく急峻)である場合には、包絡線の傾きを精度良く求めることが困難となり、利得余裕値の推定精度も低下してしまう虞がある。また、そのような傾向は、閉ループ系を伝送するサンプル信号以外の信号(音声信号又は周囲騒音信号)のレベルが高くなるほど顕著となる。 In time and, although the above embodiments have been estimated gain margin value of the closed loop by observing the slope of the envelope of the response signal when the transmitted sample signal (impulse signal or burst signals) in a closed loop system There is an upper limit to the estimated gain margin value. In other words, when the gain margin value of the closed loop system is greater than or equal to a predetermined value (the slope of the envelope of the response signal with respect to the impulse signal is steep and greater than the predetermined value), it is difficult to accurately determine the slope of the envelope. In addition, there is a possibility that the estimation accuracy of the gain margin value is also lowered. Such a tendency becomes more prominent as the level of a signal (speech signal or ambient noise signal) other than the sample signal transmitted through the closed loop system increases.

図14に示すようにマイク1、第1の増幅器2、スピーカ3、第2の増幅器4で構成される閉ループ系にインパルス信号を付加したときの応答信号の包絡線の傾きを、インパルス信号以外の信号(ランダムノイズ)を加算して観測した実験結果を図15〜図17に示す。ここで、図15は同期加算処理を行わずに20dBのランダムノイズを付加した場合、図16は同期加算処理を行って20dBのランダムノイズを付加した場合、図17は同期加算処理を行って40dBのランダムノイズを付加した場合の各応答信号波形を示しており、各図における曲線イ〜ハはそれぞれ閉ループ系の利得余裕値が3dB、6dB並びに10dBの場合を示している。これらの実験結果から、ランダムノイズ(サンプル信号以外の信号)のレベルが高くなるほど(すなわち、曲線イよりも曲線ロ、曲線ロよりも曲線ハ)、包絡線の傾きを精度良く求めることが困難になることは明らかである。   As shown in FIG. 14, the slope of the envelope of the response signal when an impulse signal is added to the closed loop system composed of the microphone 1, the first amplifier 2, the speaker 3, and the second amplifier 4 The experimental results observed by adding signals (random noise) are shown in FIGS. Here, FIG. 15 shows a case where 20 dB random noise is added without performing the synchronous addition process, FIG. 16 shows a case where 20 dB random noise is added by performing the synchronous addition process, and FIG. The response signal waveforms when the random noise is added are shown, and curves i to c in each figure show the cases where the gain margin values of the closed loop system are 3 dB, 6 dB, and 10 dB, respectively. From these experimental results, the higher the level of random noise (signals other than the sample signal) (that is, curve B rather than curve A, curve C rather than curve B), it becomes more difficult to accurately determine the slope of the envelope. Obviously.

そこで、上記課題を解決するために図1に示した構成に対して包絡線検波器11により検波される包絡線の傾きと所定値との大小関係を判定する傾き判定部26と、受話側の信号経路に挿入され傾き判定部26による判定結果に応じて増幅度を可変する利得余裕調整用増幅器27とを挿入損失量調整手段8に付加することが望ましい。 Therefore, in order to solve the above-described problem, an inclination determination unit 26 that determines the magnitude relationship between the inclination of the envelope detected by the envelope detector 11 and a predetermined value with respect to the configuration shown in FIG. It is desirable to add to the insertion loss amount adjusting means 8 a gain margin adjusting amplifier 27 that is inserted into the signal path of FIG.

図13は入損失量調整手段8を示すブロック図である。包絡線検波器11にて抽出される包絡線成分が傾き判定部26に入力され、包絡線成分の傾きが次段の閉ループ利得余裕推定部12において系の利得余裕値が精度よく推定可能な範囲か否かを傾き判定部26で判定している。また、利得余裕調整用増幅器27は第2の増幅器4の前段に挿入されており、傾き判定部26から与えられる制御信号によって増幅度が可変されるものである。 Figure 13 is a block diagram showing an insertion loss amount adjusting means 8. The envelope component extracted by the envelope detector 11 is input to the slope judgment unit 26, and the slope of the envelope component can be accurately estimated by the closed-loop gain margin estimation unit 12 at the next stage. Whether or not it is determined by the inclination determination unit 26. Further, the gain margin adjusting amplifier 27 is inserted in the previous stage of the second amplifier 4, and the degree of amplification is varied by a control signal given from the inclination determining unit 26.

而して、傾き判定部26にて包絡線検波器11にて抽出された包絡線成分の傾きを所定値と比較し、所定値よりも小さい、すなわち推定可能と判定した場合には上記包絡線成分を閉ループ利得余裕推定部12に出力するとともに利得余裕調整用増幅器27の増幅度を初期値(=1)に維持する。   Thus, the slope of the envelope component extracted by the envelope detector 11 is compared with a predetermined value by the inclination determination unit 26, and when it is determined that the inclination is smaller than the predetermined value, that is, can be estimated, the envelope The component is output to the closed-loop gain margin estimation unit 12 and the gain of the gain margin adjustment amplifier 27 is maintained at the initial value (= 1).

一方、包絡線の傾きが急峻なために、傾き判定部26にて包絡線成分の傾きが所定値よりも大きい、すなわち推定不可能と判定した場合には制御信号を出力して利得余裕調整用増幅器27の利得を増大させる。例えば、推定可能な上限値をLmax〔dB〕としたときに0〔dB〕からLmax〔dB〕刻みに段階的に増大させるようにすればよい。その結果、利得余裕調整用増幅器27の利得分だけ系の利得余裕値が減少するから、利得余裕調整用増幅器27の増幅度を適当な値に設定すれば、系の利得余裕を閉ループ利得余裕推定部12における推定が可能な値とすることができる。また、これ以降は包絡線検波器11で抽出される包絡線成分の傾きが小さくなって傾き判定部26にて推定可能と判定されるようになり、上記包絡線成分が閉ループ利得余裕推定部12に入力されて利得余裕の推定が行われる。 On the other hand, since the slope of the envelope curve is steep, when the slope determination unit 26 determines that the slope of the envelope component is larger than a predetermined value, that is, cannot be estimated, a control signal is output to adjust the gain margin. The gain of the amplifier 27 is increased. For example, when the estimable upper limit value is L max [dB], it may be increased in steps from 0 [dB] to L max [dB]. As a result, the gain margin value of the system is reduced by the gain of the gain margin adjusting amplifier 27. Therefore, if the gain of the gain margin adjusting amplifier 27 is set to an appropriate value, the gain margin of the system is estimated as a closed loop gain margin. The value can be estimated by the unit 12. Thereafter, the inclination of the envelope component extracted by the envelope detector 11 becomes small and it is determined that the inclination can be estimated by the inclination determination unit 26, and the envelope component is determined as the closed loop gain margin estimation unit 12. The gain margin is estimated.

ここで、閉ループ利得余裕推定部12にて推定される利得余裕値に利得余裕調整用増幅器27の増幅度を乗じた値が実際の系の利得余裕値となるから、この値に基づいて挿入損失量の調整を行う必要がある。   Here, since a value obtained by multiplying the gain margin value estimated by the closed-loop gain margin estimation unit 12 by the gain of the gain margin adjustment amplifier 27 becomes the actual gain margin value, the insertion loss is based on this value. It is necessary to adjust the amount.

上述のように包絡線検波器11により検波される包絡線の傾きと所定値との大小関係を判定する傾き判定部26と、受話側の信号経路に挿入され傾き判定部26による判定結果に応じて増幅度を可変する利得余裕調整用増幅器27とを挿入損失量調整手段8に具備すれば、閉ループ系の利得余裕が大きい状態においても利得余裕値を精度よく推定することができる。その結果、閉ループ系の利得余裕値を常に充分な値に維持可能なように挿入損失量を調整することができる。なお、上記構成では利得余裕調整用増幅器27を送話側の信号経路に挿入しているが受話側の信号経路に挿入しても同様の作用効果を奏することは言うまでもない。また、上記構成では図1に示した構成に対して傾き判定部26及び利得余裕調整用増幅器27を付加する構成としているが、図4又は図5に示した構成あるいは図18に示す基本構成に対して傾き判定部26及び利得余裕調整用増幅器27を付加する構成としても同様の作用効果を奏する。 As described above, the determined inclination judgment unit 26 the magnitude relation between the slope and the predetermined value of the envelope is detected by an envelope detector 11, the determination result by the inclination determination unit 26 is inserted in the signal path of the receiving side if and a gain margin adjusting amplifier 27 for changing the amplification factor in response to the insertion loss adjusting means 8 can also estimate the gain margin value precisely in the gain margin is large state of the closed loop system. As a result, the amount of insertion loss can be adjusted so that the gain margin value of the closed loop system can always be maintained at a sufficient value. In the above configuration , the gain margin adjusting amplifier 27 is inserted in the signal path on the transmission side, but it goes without saying that the same effect can be obtained even if it is inserted in the signal path on the reception side. Further, in the above configuration , the inclination determination unit 26 and the gain margin adjustment amplifier 27 are added to the configuration shown in FIG. 1, but the configuration shown in FIG. 4 or FIG. 5 or the basic configuration shown in FIG. On the other hand, the same effect can be obtained by adding the inclination determining unit 26 and the gain margin adjusting amplifier 27.

なお、上記実施形態1〜における各挿入損失量調整手段8は、基本構成と同様にマイコンやDSPを用いて実現可能である。また、サンプル信号としてインパルス信号を例示したが、バースト信号をサンプル信号に用いてもよいことは言うまでもない。 In addition, each insertion loss amount adjusting means 8 in the first to sixth embodiments can be realized by using a microcomputer or a DSP similarly to the basic configuration. Although the impulse signal is exemplified as the sample signal, it is needless to say that a burst signal may be used as the sample signal.

実施形態における挿入損失量調整手段の主要構成を示すブロック図である。It is a block diagram which shows the main structures of the insertion loss amount adjustment means in embodiment. 同上における同期制御部並びに同期加算平均部を示すブロック図である。It is a block diagram which shows the synchronous control part and synchronous addition average part in the same as the above. 同上の動作を説明するための説明図である。It is explanatory drawing for demonstrating operation | movement same as the above. 実施形態における挿入損失量調整手段の他の主要構成を示すブロック図である。It is a block diagram which shows the other main structure of the insertion loss amount adjustment means in embodiment. 実施形態における挿入損失量調整手段のさらに他の主要構成を示すブロック図である。It is a block diagram which shows the further another main structure of the insertion loss amount adjustment means in embodiment. 実施形態1における挿入損失量調整手段を示すブロック図である。It is a block diagram which shows the insertion loss amount adjustment means in Embodiment 1. 実施形態2における挿入損失量調整手段を示すブロック図である。It is a block diagram which shows the insertion loss amount adjustment means in Embodiment 2. 実施形態3における挿入損失量調整手段を示すブロック図である。It is a block diagram which shows the insertion loss amount adjustment means in Embodiment 3. 実施形態4における挿入損失量調整手段を示すブロック図である。It is a block diagram which shows the insertion loss amount adjustment means in Embodiment 4. 実施形態5における挿入損失量調整手段を示すブロック図である。It is a block diagram which shows the insertion loss amount adjustment means in Embodiment 5. 実施形態6における挿入損失量調整手段を示すブロック図である。It is a block diagram which shows the insertion loss amount adjustment means in Embodiment 6. 同上における周囲騒音抽出部を示すブロック図である。It is a block diagram which shows the ambient noise extraction part in the same as the above. 実施形態1〜6における挿入損失量調整手段の別の構成を示すブロック図である。It is a block diagram which shows another structure of the insertion loss amount adjustment means in Embodiments 1-6 . 同上の説明図である。It is explanatory drawing same as the above. 同上の説明図である。It is explanatory drawing same as the above. 同上の説明図である。It is explanatory drawing same as the above. 同上の説明図である。It is explanatory drawing same as the above. (a)は本発明の基本構成を示すブロック図、(b)は同じく挿入損失量調整手段の基本構成を示すブロック図である。(A) is a block diagram showing the basic configuration of the present invention, (b) is a block diagram showing the basic configuration of the insertion loss amount adjusting means. 同上における閉ループを説明するための説明図である。It is explanatory drawing for demonstrating the closed loop in the same as the above.

符号の説明Explanation of symbols

8 挿入損失量調整手段
9 インパルス信号発生器
10 加算器
11 包絡線検波器
12 閉ループ利得余裕推定部
15 同期制御部
16 同期加算平均部
17 収束判定部
DESCRIPTION OF SYMBOLS 8 Insertion loss amount adjustment means 9 Impulse signal generator 10 Adder 11 Envelope detector 12 Closed loop gain margin estimation part 15 Synchronization control part 16 Synchronization addition average part 17 Convergence determination part

Claims (10)

集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する信号の自己相関関数を算出する自己相関関数算出部とを具備し、同期制御部は、自己相関関数の変数である時間シフト量と、所定値以上の雑音成分の抑圧量を得るために必要な同期加算回数との関係を、自己相関関数算出部により算出される自己相関関数と照合することにより、同期加算処理に要する時間を最小化するために必要なサンプル信号の送出間隔及び同期加算回数を求めて当該送出間隔及び同期加算回数を調整して成ることを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion loss that estimates the gain margin in the closed loop formed through the microphone and speaker according to the control signal and the response signal to the sample signal sent to the signal path and adjusts the loss amount based on the estimation result via the control means And an insertion loss amount adjusting means for sending a sample signal to the signal path and a sample signal for the audio signal. In the closed loop based on the output signal of the envelope detector, and the envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path. A closed loop gain margin estimation unit for estimating a gain margin, a synchronization control unit for performing timing control for sending a sample signal from a sample signal generator to a signal path a plurality of times at predetermined time intervals, and a front stage of an envelope detector A synchronization addition averaging unit that receives the timing information from the synchronization control unit and performs a synchronous addition averaging process on the response signal with respect to the sample signal; and an autocorrelation function calculation unit that calculates an autocorrelation function of the signal existing in the closed loop ; comprising a synchronization control unit, and time shift amount is a variable of the autocorrelation function, synchronization necessary to obtain the suppression amount of the predetermined value or more of the noise component By comparing the relationship with the number of calculations with the autocorrelation function calculated by the autocorrelation function calculation unit, the sample signal transmission interval and the number of synchronization additions required to minimize the time required for the synchronous addition process are obtained. The loudspeaker is characterized by adjusting the transmission interval and the number of synchronous additions . 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion loss that estimates the gain margin in the closed loop formed through the microphone and speaker according to the control signal and the response signal to the sample signal sent to the signal path and adjusts the loss amount based on the estimation result via the control means And an insertion loss amount adjusting means for sending a sample signal to the signal path and a sample signal for the audio signal. In the closed loop based on the output signal of the envelope detector, and the envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path. A closed loop gain margin estimation unit for estimating a gain margin, a synchronization control unit for performing timing control for sending a sample signal from a sample signal generator to a signal path a plurality of times at predetermined time intervals, and a front stage of an envelope detector A synchronous addition averaging unit that receives the timing information from the synchronization control unit and performs synchronous addition averaging processing on the response signal to the sample signal, a noise level estimation unit that estimates the level of ambient noise existing in the closed loop, and noise A synchronization addition number adjustment unit that adjusts the number of synchronization additions in the synchronization addition averaging unit based on the estimation result by the level estimation unit. The number-of-addition adjustment unit is a delay associated with the synchronous addition average with respect to the ambient noise level estimated by the noise level estimation unit based on the statistical properties of the ambient noise level and the number of synchronous additions that have been quantitatively examined in advance. expanding voice call machine characterized by comprising setting the minimum synchronous addition number required to minimize the time. 閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部を挿入損失量調整手段が具備し、抽出された周囲騒音信号から騒音レベル推定部が閉ループ中に存在する周囲騒音のレベルを推定して成ることを特徴とする請求項2記載の拡声通話機。 The insertion loss adjustment means includes an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and the noise level estimation unit determines the ambient noise level existing in the closed loop from the extracted ambient noise signal. The loudspeaker according to claim 2, wherein the loudspeaker is estimated . 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいてサンプル信号の送出レベルを調整する送出レベル調整部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部とを具備し、騒音レベル推定部は、周囲騒音抽出部で抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定して成り、送出レベル調整部は、閉ループ中の周囲騒音レベルに対するサンプル信号送出レベルの比が、閉ループにサンプル信号が送出される以前の周囲騒音レベルに対するサンプル信号送出レベルの前記比に対して所定値以上となるようにサンプル信号の送出レベルを調整することを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion loss that estimates the gain margin in the closed loop formed through the microphone and speaker according to the control signal and the response signal to the sample signal sent to the signal path and adjusts the loss amount based on the estimation result via the control means And an insertion loss amount adjusting means for sending a sample signal to the signal path and a sample signal for the audio signal. In the closed loop based on the output signal of the envelope detector, and the envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path. A closed loop gain margin estimation unit for estimating a gain margin, a synchronization control unit for performing timing control for sending a sample signal from a sample signal generator to a signal path a plurality of times at predetermined time intervals, and a front stage of an envelope detector A synchronous addition averaging unit that receives the timing information from the synchronization control unit and performs synchronous addition averaging processing on the response signal to the sample signal, a noise level estimation unit that estimates the level of ambient noise existing in the closed loop, and noise A transmission level adjustment unit that adjusts the transmission level of the sample signal based on the estimation result by the level estimation unit, and is transmitted during the closed loop An ambient noise extraction unit for extracting an ambient noise signal from the signal, the noise level estimation unit is configured by estimating the level of ambient noise existing in the closed loop from the ambient noise signal extracted by the ambient noise extraction unit, In the transmission level adjusting unit, the ratio of the sample signal transmission level to the ambient noise level in the closed loop is a predetermined value or more with respect to the ratio of the sample signal transmission level to the ambient noise level before the sample signal is transmitted to the closed loop. The loudspeaker is characterized by adjusting the transmission level of the sample signal as described above . 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出したサンプル信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路にサンプル信号を送出するサンプル信号発生器と、音声信号にサンプル信号を加算する加算器と、サンプル信号を信号経路に加算したときに加算器から出力される前記応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔でサンプル信号発生器から信号経路へ複数回にわたってサンプル信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けてサンプル信号に対する前記応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部と、抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいてサンプル信号の送出レベルを調整する送出レベル調整部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、送出レベル調整部は、閉ループ中の周囲騒音レベルに対するサンプル信号送出レベルの比が、閉ループにサンプル信号が送出される以前の周囲騒音レベルに対するサンプル信号送出レベルの前記比が所定値以上となるようにサンプル信号の送出レベルを調整して成り、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion loss that estimates the gain margin in the closed loop formed through the microphone and speaker according to the control signal and the response signal to the sample signal sent to the signal path and adjusts the loss amount based on the estimation result via the control means And an insertion loss amount adjusting means for sending a sample signal to the signal path and a sample signal for the audio signal. In the closed loop based on the output signal of the envelope detector, and the envelope detector for detecting the envelope of the response signal output from the adder when the sample signal is added to the signal path. A closed loop gain margin estimation unit for estimating a gain margin, a synchronization control unit for performing timing control for sending a sample signal from a sample signal generator to a signal path a plurality of times at predetermined time intervals, and a front stage of an envelope detector A synchronous addition averaging unit that receives timing information from the synchronization control unit and performs synchronous addition averaging processing on the response signal with respect to the sample signal, and an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during a closed loop A noise level estimator that estimates the level of ambient noise existing in the closed loop from the extracted ambient noise signal, and a noise level estimator. A transmission level adjustment unit that adjusts the transmission level of the sample signal based on the estimation result, and a synchronous addition number adjustment unit that adjusts the number of synchronous additions in the synchronous addition averaging unit based on the estimation result by the noise level estimation unit, The transmission level adjustment unit adjusts the ratio of the sample signal transmission level to the ambient noise level in the closed loop so that the ratio of the sample signal transmission level to the ambient noise level before the sample signal is transmitted to the closed loop is equal to or greater than a predetermined value. The synchronization addition number adjusting unit is estimated by the noise level estimation unit based on the statistical properties of the ambient noise level and the number of synchronization additions that have been quantitatively examined in advance. Set the minimum number of synchronous additions necessary to minimize the delay time associated with the synchronous average for the ambient noise level Expanding voice call machine said. 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する信号の自己相関関数を算出する自己相関関数算出部とを具備し、同期制御部は、自己相関関数の変数である時間シフト量と、所定値以上の雑音成分の抑圧量を得るために必要な同期加算回数との関係を、自己相関関数算出部により算出される自己相関関数と照合することにより、同期加算処理に要する時間を最小化するために必要な疑似白色信号発生器からの疑似白色信号の送出間隔及び同期加算回数を求めて当該送出間隔及び同期加算回数を調整することを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion that estimates the gain margin in the closed loop formed through the microphone and the speaker and adjusts the loss amount based on the estimation result via the control means according to the control means and the response signal to the pseudo white signal sent to the signal path Loss amount adjusting means, and the insertion loss amount adjusting means includes a pseudo white signal generator for inserting a pseudo white signal having a finite period in the signal path, and an audio signal. An adder that adds a pseudo white signal, and a cross correlation value of the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path, and based on the obtained cross correlation value A cross-correlation operation unit that estimates the impulse response of the closed loop, an envelope detector that detects an envelope of the impulse response signal of the closed loop estimated by the cross-correlation operation unit, and a closed loop based on the output signal of the envelope detector A closed loop gain margin estimating unit for estimating a gain margin, a synchronous control unit for performing timing control for sending a pseudo white signal to a signal path from a pseudo white signal generator to a signal path at a predetermined time interval, and an envelope detector The closed-loop impulse response signal estimated by the cross-correlation calculation unit in response to timing information from the synchronization control unit A synchronous addition averaging unit that performs a periodical addition averaging process, and an autocorrelation function calculation unit that calculates an autocorrelation function of a signal existing in a closed loop, and the synchronization control unit includes a time shift amount that is a variable of the autocorrelation function, The time required for synchronous addition processing is minimized by checking the relationship between the number of synchronous additions necessary to obtain a noise component suppression amount equal to or greater than a predetermined value and the autocorrelation function calculated by the autocorrelation function calculation unit. A loudspeaker, wherein a transmission interval and the number of synchronous additions of a pseudo white signal from a pseudo white signal generator required for conversion are obtained, and the transmission interval and the number of synchronous additions are adjusted . 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion that estimates the gain margin in the closed loop formed through the microphone and the speaker and adjusts the loss amount based on the estimation result via the control means according to the control means and the response signal to the pseudo white signal sent to the signal path Loss amount adjusting means, and the insertion loss amount adjusting means includes a pseudo white signal generator for inserting a pseudo white signal having a finite period in the signal path, and an audio signal. An adder that adds a pseudo white signal, and a cross correlation value of the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path, and based on the obtained cross correlation value A cross-correlation operation unit that estimates the impulse response of the closed loop, an envelope detector that detects an envelope of the impulse response signal of the closed loop estimated by the cross-correlation operation unit, and a closed loop based on the output signal of the envelope detector A closed loop gain margin estimating unit for estimating a gain margin, a synchronous control unit for performing timing control for sending a pseudo white signal to a signal path from a pseudo white signal generator to a signal path at a predetermined time interval, and an envelope detector The closed-loop impulse response signal estimated by the cross-correlation calculation unit in response to timing information from the synchronization control unit The synchronous addition averaging unit for performing the periodic addition averaging process, the noise level estimation unit for estimating the level of ambient noise existing in the closed loop, and the number of synchronous additions in the synchronous addition averaging unit based on the estimation result by the noise level estimation unit A synchronization addition number adjustment unit, and the synchronization addition number adjustment unit is configured to estimate the ambient level estimated by the noise level estimation unit based on the statistical properties of the ambient noise level and the number of synchronization additions that have been quantitatively examined in advance. A loudspeaker having a minimum number of synchronization additions required for minimizing a delay time associated with a synchronization addition average with respect to a noise level . 閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部を挿入損失量調整手段が具備し、抽出された周囲騒音信号から騒音レベル推定部が閉ループ中に存在する周囲騒音のレベルを推定して成ることを特徴とする請求項7記載の拡声通話機。 The insertion loss adjustment means includes an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and the noise level estimation unit determines the ambient noise level existing in the closed loop from the extracted ambient noise signal. 8. The loudspeaker according to claim 7, wherein the loudspeaker is estimated . 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて疑似白色信号の送出レベルを調整する送出レベル調整部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部とを具備し、騒音レベル推定部は、周囲騒音抽出部で抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定して成り、送出レベル調整部は、閉ループ中の周囲騒音レベルに対する擬似白色信号送出レベルの比が、閉ループに擬似白色信号が送出される以前の周囲騒音レベルに対する擬似白色信号送出レベルの前記比に対して所定値以上となるように擬似白色信号の送出レベルを調整することを特徴とする拡声通話機。 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion that estimates the gain margin in the closed loop formed through the microphone and the speaker and adjusts the loss amount based on the estimation result via the control means according to the control means and the response signal to the pseudo white signal sent to the signal path Loss amount adjusting means, and the insertion loss amount adjusting means includes a pseudo white signal generator for inserting a pseudo white signal having a finite period in the signal path, and an audio signal. An adder that adds a pseudo white signal, and a cross correlation value of the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path, and based on the obtained cross correlation value A cross-correlation operation unit that estimates the impulse response of the closed loop, an envelope detector that detects an envelope of the impulse response signal of the closed loop estimated by the cross-correlation operation unit, and a closed loop based on the output signal of the envelope detector A closed loop gain margin estimating unit for estimating a gain margin, a synchronous control unit for performing timing control for sending a pseudo white signal to a signal path from a pseudo white signal generator to a signal path at a predetermined time interval, and an envelope detector The closed-loop impulse response signal estimated by the cross-correlation calculation unit in response to timing information from the synchronization control unit Synchronous addition averaging unit that performs periodic addition averaging processing, noise level estimation unit that estimates the level of ambient noise existing in the closed loop, and transmission level that adjusts the transmission level of the pseudo white signal based on the estimation result by the noise level estimation unit An adjustment unit and an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and the noise level estimation unit exists in the closed loop from the ambient noise signal extracted by the ambient noise extraction unit The transmission level adjustment unit is configured by estimating the ambient noise level, and the ratio of the pseudo white signal transmission level to the ambient noise level in the closed loop is the pseudo white signal with respect to the ambient noise level before the pseudo white signal is transmitted to the closed loop. A loudspeaker , wherein a transmission level of a pseudo white signal is adjusted to be a predetermined value or more with respect to the ratio of the transmission levels . 集音した音を送話側の音声信号として出力するマイクロホンと、マイクロホンからの音声信号を増幅する第1の増幅手段と、受話側の音声信号に応じて鳴動するスピーカと、スピーカへ出力される音声信号を増幅する第2の増幅手段と、送話側及び受話側の少なくとも一方の信号経路に所定量の損失を挿入する損失挿入手段と、損失挿入手段から挿入される損失量を可変制御する制御手段と、信号経路に送出した疑似白色信号に対する応答信号に応じて、マイクロホン及びスピーカを通じて形成される閉ループにおける利得余裕を推定するとともに制御手段を介して推定結果に基づく損失量の調整を行う挿入損失量調整手段とを備え、挿入損失量調整手段が、信号経路に周期が有限の疑似白色信号を挿入する疑似白色信号発生器と、音声信号に疑似白色信号を加算する加算器と、疑似白色信号と当該疑似白色信号を信号経路に加算したときに加算器から出力される前記応答信号の相互相関値を求めるとともに求めた相互相関値に基づいて閉ループのインパルス応答を推定する相互相関演算部と、相互相関演算部により推定された閉ループのインパルス応答信号の包絡線を検波する包絡線検波器と、包絡線検波器の出力信号に基づいて閉ループにおける利得余裕を推定する閉ループ利得余裕推定部と、所定の時間間隔で疑似白色信号発生器から信号経路へ複数回にわたって疑似白色信号を送出させるためのタイミング制御を行う同期制御部と、包絡線検波器の前段に設けられ同期制御部からのタイミング情報を受けて相互相関演算部により推定された閉ループのインパルス応答信号を同期加算平均処理する同期加算平均部と、閉ループ中に伝送される信号から周囲騒音信号を抽出する周囲騒音抽出部と、抽出された周囲騒音信号から閉ループ中に存在する周囲騒音のレベルを推定する騒音レベル推定部と、騒音レベル推定部による推定結果に基づいて疑似白色信号の送出レベルを調整する送出レベル調整部と、騒音レベル推定部による推定結果に基づいて同期加算平均部における同期加算回数を調整する同期加算回数調整部とを具備し、送出レベル調整部は、閉ループ中の周囲騒音レベルに対するサンプル信号送出レベルの比が、閉ループにサンプル信号が送出される以前の周囲騒音レベルに対するサンプル信号送出レベルの前記比が所定値以上となるようにサンプル信号の送出レベルを調整して成り、同期加算回数調整部は、予め定量的に調べられている周囲騒音レベルと同期加算回数との統計的性質に基づいて、騒音レベル推定部で推定される周囲騒音レベルに対して同期加算平均に伴う遅延時間を最小化するために必要な最小限の同期加算回数を設定して成ることを特徴とする拡声通話機 A microphone that outputs the collected sound as an audio signal on the transmitting side, a first amplifying means that amplifies the audio signal from the microphone, a speaker that rings according to the audio signal on the receiving side, and an output to the speaker A second amplifying means for amplifying the audio signal; a loss inserting means for inserting a predetermined amount of loss into at least one of the signal paths on the transmitting side and the receiving side; and variably controlling a loss amount inserted from the loss inserting means. Insertion that estimates the gain margin in the closed loop formed through the microphone and the speaker and adjusts the loss amount based on the estimation result via the control means according to the control means and the response signal to the pseudo white signal sent to the signal path Loss amount adjusting means, and the insertion loss amount adjusting means includes a pseudo white signal generator for inserting a pseudo white signal having a finite period in the signal path, and an audio signal. An adder that adds a pseudo white signal, and a cross correlation value of the response signal output from the adder when the pseudo white signal and the pseudo white signal are added to the signal path, and based on the obtained cross correlation value A cross-correlation operation unit that estimates the impulse response of the closed loop, an envelope detector that detects an envelope of the impulse response signal of the closed loop estimated by the cross-correlation operation unit, and a closed loop based on the output signal of the envelope detector A closed loop gain margin estimating unit for estimating a gain margin, a synchronous control unit for performing timing control for sending a pseudo white signal to a signal path from a pseudo white signal generator to a signal path at a predetermined time interval, and an envelope detector The closed-loop impulse response signal estimated by the cross-correlation calculation unit in response to timing information from the synchronization control unit A synchronous addition averaging unit that performs periodic addition averaging processing, an ambient noise extraction unit that extracts an ambient noise signal from a signal transmitted during the closed loop, and an ambient noise level that exists in the closed loop is estimated from the extracted ambient noise signal The noise level estimation unit, the transmission level adjustment unit for adjusting the transmission level of the pseudo white signal based on the estimation result by the noise level estimation unit, and the number of synchronization additions in the synchronization addition averaging unit based on the estimation result by the noise level estimation unit And a sending level adjusting unit that sends the sample signal to the ambient noise level before the sample signal is sent to the closed loop, with the ratio of the sample signal sending level to the ambient noise level in the closed loop being adjusted. It is made by adjusting the transmission level of the sample signal so that the ratio of the levels becomes a predetermined value or more. Based on the statistical properties of the ambient noise level and the number of synchronization additions that have been quantitatively examined in advance, the delay time associated with the synchronization addition average is minimized with respect to the ambient noise level estimated by the noise level estimation unit. A loudspeaker having a minimum number of synchronous additions necessary for the purpose .
JP2005030869A 1998-03-16 2005-02-07 Loudspeaker Expired - Fee Related JP4349296B2 (en)

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