JP2000270089A - Amplified sound speech device - Google Patents
Amplified sound speech deviceInfo
- Publication number
- JP2000270089A JP2000270089A JP11072927A JP7292799A JP2000270089A JP 2000270089 A JP2000270089 A JP 2000270089A JP 11072927 A JP11072927 A JP 11072927A JP 7292799 A JP7292799 A JP 7292799A JP 2000270089 A JP2000270089 A JP 2000270089A
- Authority
- JP
- Japan
- Prior art keywords
- unit
- loss
- feedback gain
- output
- value
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Granted
Links
- 238000003780 insertion Methods 0.000 claims abstract description 97
- 230000037431 insertion Effects 0.000 claims abstract description 97
- 230000005540 biological transmission Effects 0.000 claims abstract description 54
- 230000005236 sound signal Effects 0.000 claims description 23
- 238000004891 communication Methods 0.000 claims description 19
- 238000012545 processing Methods 0.000 claims description 19
- 230000003044 adaptive effect Effects 0.000 claims description 13
- 230000004044 response Effects 0.000 claims description 8
- 238000000034 method Methods 0.000 claims description 4
- 230000008569 process Effects 0.000 claims description 3
- 230000001934 delay Effects 0.000 claims description 2
- 230000003321 amplification Effects 0.000 claims 2
- 238000003199 nucleic acid amplification method Methods 0.000 claims 2
- 230000003014 reinforcing effect Effects 0.000 abstract 1
- 230000000694 effects Effects 0.000 description 9
- 238000010586 diagram Methods 0.000 description 8
- 238000009499 grossing Methods 0.000 description 3
- 230000008901 benefit Effects 0.000 description 2
- 230000008859 change Effects 0.000 description 2
- 238000006243 chemical reaction Methods 0.000 description 1
- 230000003111 delayed effect Effects 0.000 description 1
- 238000012544 monitoring process Methods 0.000 description 1
- 230000007704 transition Effects 0.000 description 1
Classifications
-
- Y—GENERAL TAGGING OF NEW TECHNOLOGICAL DEVELOPMENTS; GENERAL TAGGING OF CROSS-SECTIONAL TECHNOLOGIES SPANNING OVER SEVERAL SECTIONS OF THE IPC; TECHNICAL SUBJECTS COVERED BY FORMER USPC CROSS-REFERENCE ART COLLECTIONS [XRACs] AND DIGESTS
- Y02—TECHNOLOGIES OR APPLICATIONS FOR MITIGATION OR ADAPTATION AGAINST CLIMATE CHANGE
- Y02D—CLIMATE CHANGE MITIGATION TECHNOLOGIES IN INFORMATION AND COMMUNICATION TECHNOLOGIES [ICT], I.E. INFORMATION AND COMMUNICATION TECHNOLOGIES AIMING AT THE REDUCTION OF THEIR OWN ENERGY USE
- Y02D30/00—Reducing energy consumption in communication networks
- Y02D30/70—Reducing energy consumption in communication networks in wireless communication networks
Landscapes
- Interconnected Communication Systems, Intercoms, And Interphones (AREA)
- Cable Transmission Systems, Equalization Of Radio And Reduction Of Echo (AREA)
- Telephone Function (AREA)
Abstract
Description
【０００１】[0001]
【発明の属する技術分野】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装置に関するもので
ある。BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a voice communication device used in homes, buildings, factories, and the like.
【０００２】[0002]
【従来の技術】従来より、インターホンや電話機あるい
はＰＨＳ等の拡声通話装置においては、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という問題があり、ハウリング
が生じた場合には通話を継続することが困難となるた
め、ハウリングを抑圧する手段が必要不可欠となってい
た。2. Description of the Related Art Conventionally, in a loudspeaker apparatus such as an intercom, a telephone or a PHS, a closed loop is formed due to acoustic feedback from a speaker to a microphone and impedance mismatch in a hybrid circuit (2-4 wire conversion circuit). If the gain of the closed loop becomes 1 or more due to the reason that the gain of the amplifier is too large or the like, there is a problem that howling occurs. If the howling occurs, it becomes difficult to continue the communication, and the howling is suppressed. Means were indispensable.
【０００３】そこで従来は、送受話信号を監視すること
で通話状態が受話状態か送話状態の何れであるかを判別
し、受話状態の時には送話信号に対して所定の損失量を
挿入し、送話状態の時は受話信号に対して所定の損失量
を挿入することにより、閉ループの一巡伝達利得を低減
させる、所謂音声スイッチと呼ばれるものが広く用いら
れてきた。この音声スイッチを用いる場合、挿入損失量
を大きくしすぎると通話中に切断感を生じる等の通話品
質の劣化を招くため、挿入損失量は所望の利得余裕（ハ
ウリングマージン）が得られるだけの必要最小限とする
ことが望ましい。Conventionally, therefore, it is determined whether the communication state is a reception state or a transmission state by monitoring a transmission / reception signal. In the reception state, a predetermined loss is inserted into the transmission signal. In a transmission state, a so-called voice switch has been widely used in which a predetermined amount of loss is inserted into a received signal to reduce a closed loop loop transmission gain. In the case of using this voice switch, if the insertion loss is too large, the quality of the call such as a feeling of disconnection during a call is deteriorated, so that the insertion loss needs to be sufficient to obtain a desired gain margin (howling margin). It is desirable to minimize it.
【０００４】上述のような課題に対して、近端側（音声
スイッチの挿入位置よりも端末側）と遠端側（音声スイ
ッチの挿入位置よりも回線側）の帰還利得を随時推定
し、その推定結果に応じて挿入損失量を適応的に制御す
ることによって、常に一定の利得余裕を保ちながら挿入
損失量を必要最小限とする音声スイッチが提供されてい
る（特開昭６３−２１２２５１号公報等参照）。In order to solve the above problem, feedback gains on the near end (terminal side from the position where the voice switch is inserted) and the far end (line side from the position where the voice switch is inserted) are estimated as needed. A voice switch that minimizes the insertion loss while maintaining a constant gain margin by adaptively controlling the insertion loss according to the estimation result has been provided (Japanese Patent Application Laid-Open No. Sho 63-212251). Etc.).
【０００５】上記公報に記載されている従来技術の原理
を図６を参照して説明する。近端側のマイクロホン１に
送話減衰器３０が接続されるとともにスピーカ３に受話
減衰器３１が接続されているとき、受話減衰器３１の入
出力信号ｒin，ｒoutを整流平滑して求められる振幅値
をＲｉ，Ｒｏ、送話減衰器３０の入出力信号ｓin，ｓ
outを整流平滑して求められる振幅値をＳｉ，Ｓｏと定
義する。そして、近端側の帰還利得αを推定するために
送話減衰器３０の入力信号ｓinの振幅値Ｓｉ、受話減衰
器３１の出力信号ｒoutの振幅値Ｒｏの比Ｓｉ／Ｒｏの
受話状態における観測値を用いており、遠端側の帰還利
得βを推定するために受話減衰器３１の入力信号ｒinの
振幅値Ｒｉ、送話減衰器３０の出力信号ｓoutの振幅値
Ｓｏの比Ｒｉ／Ｓｏの送話状態における観測値を用いて
いる。つまり上記公報に記載されているものは、近端側
話者の音声信号が存在しない場合、すなわち図６におけ
るＶｓ＝０のときはＳｉ／Ｒｏ＝αとなり、遠端側話者
の音声信号が存在しない場合、すなわち図６におけるＶ
ｒ＝０のときはＲｉ／Ｓｏ＝βとなるという原理に基づ
いている。The principle of the prior art described in the above publication
Will be described with reference to FIG. To microphone 1 on the near end
The transmission attenuator 30 is connected and the speaker 3 receives a call.
When the attenuator 31 is connected, the reception attenuator 31
Output signal rin, RoutValue obtained by rectifying and smoothing
Are the input and output signals s of the transmission attenuator 30.in, S
outThe amplitude values obtained by rectifying and smoothing are defined as Si and So.
Justify. Then, in order to estimate the feedback gain α on the near end side,
Input signal s of transmission attenuator 30inAmplitude value Si, reception attenuation
Output signal r of the unit 31outOf the amplitude value Ro of the ratio Si / Ro
The observation value in the receiving state is used, and the return
The input signal r of the receiving attenuator 31 for estimating the gain βinof
Amplitude value Ri, output signal s of transmission attenuator 30outAmplitude value
Using the observed value in the transmitting state of the ratio Ri / So of So
I have. In other words, what is described in the above publication is the near end side
When there is no voice signal of the speaker, that is, in FIG.
When Vs = 0, Si / Ro = α, and the far end speaker
Is not present, that is, V in FIG.
Based on the principle that Ri / So = β when r = 0
Have been.
【０００６】[0006]
【発明が解決しようとする課題】上記従来例において
は、時刻ｔにおいて求められる受話減衰器３１の入出力
信号ｒin，ｒoutの振幅値をＲi(t)，Ｒo(t)、送話減衰
器３０の入出力信号ｓin，ｓoutの振幅値Ｓi(t)，Ｓo
(t)とするとＳi(t)／Ｒo(t)を用いて近端側の帰還利得
αを推定し、Ｒi(t)／Ｓo(t)を用いて遠端側の帰還利得
βを推定している。しかし、実際の帰還経路には伝達遅
延時間があり、近端側及び遠端側の各帰還経路の伝達遅
延時間を各々Ｔα，Ｔβとすると、近端側の帰還利得α
はＶｒ＝０におけるＳi(t)／Ｒo(t-Ｔα)で表され、遠
端側の帰還利得βはＶｓ＝０におけるＲi(t)／Ｓo(t-Ｔ
β)で表される。In the above conventional example, the amplitude values of the input / output signals r in and r out of the reception attenuator 31 obtained at time t are represented by Ri (t) and Ro (t), and the transmission attenuation. Values Si (t), So of the input / output signals s in , s out of the detector 30
(t), the feedback gain α at the near end is estimated using Si (t) / Ro (t), and the feedback gain β at the far end is estimated using Ri (t) / So (t). ing. However, an actual feedback path has a transmission delay time. If the transmission delay times of the near-end and far-end feedback paths are Tα and Tβ, respectively, the near-end feedback gain α
Is expressed as Si (t) / Ro (t-Tα) at Vr = 0, and the feedback gain β at the far end is Ri (t) / So (t-Tα at Vs = 0.
β).
【０００７】而して、上記従来例では帰還利得α，βを
推定する際に経路の伝達遅延時間Ｔα，Ｔβを考慮して
いないため、これら伝達遅延時間Ｔα，Ｔβの値が無視
できないほど長いときには真の帰還利得と推定値との誤
差が大きくなってしまう。従って、そのような場合には
信号経路に挿入すべき損失量を精度よく算出することが
できないと言う問題がある。In the above conventional example, since the propagation delay times Tα and Tβ of the path are not taken into account when estimating the feedback gains α and β, the values of these propagation delay times Tα and Tβ are too long to be ignored. Sometimes, the error between the true feedback gain and the estimated value increases. Therefore, in such a case, there is a problem that the loss amount to be inserted into the signal path cannot be calculated with high accuracy.
【０００８】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、高い精度で帰還利得を
推定し挿入損失量を適切な値に制御することで同時通話
性に優れた拡声通話装置を提供することにある。SUMMARY OF THE INVENTION The present invention has been made in view of the above-mentioned problems, and has as its object to provide excellent simultaneous voice communication by estimating a feedback gain with high accuracy and controlling an insertion loss to an appropriate value. Loudspeaker apparatus.
【０００９】[0009]
【課題を解決するための手段】上記目的を達成するため
に、請求項１の発明は、集音した音を送話側の音声信号
として出力するマイクロホンと、マイクロホンからの音
声信号を増幅する第１の増幅手段と、受話側の音声信号
に応じて鳴動するスピーカと、スピーカへ出力される音
声信号を増幅する第２の増幅手段と、送話側の信号経路
に所定量の損失を挿入する送話側損失挿入手段と、受話
側の信号経路に所定量の損失を挿入する受話側損失挿入
手段と、受話側損失挿入手段及び送話側損失挿入手段の
挿入損失量を調整する挿入損失量調整手段とを備えた拡
声通話装置において、挿入損失量調整手段は、送話側並
びに受話側の音声信号を監視して通話状態を判定する通
話状態判定部と、送話側損失挿入手段の入力信号並びに
受話側損失挿入手段の出力信号から近端側の帰還利得を
推定する近端側帰還利得推定部と、受話側損失挿入手段
の入力信号並びに送話側損失挿入手段の出力信号から遠
端側の帰還利得を推定する遠端側帰還利得推定部と、近
端側帰還利得推定部及び遠端側帰還利得推定部の各推定
値に基づいて閉ループに挿入すべき総損失量を算出する
総損失量算出部と、通話状態判定部の判定結果と総損失
量算出部の算出値に応じて送話側損失挿入手段及び受話
側挿入損失手段の各挿入損失量を決定する挿入損失量分
配処理部とを具備し、近端側帰還利得推定部は、送話側
損失挿入手段の入力信号の時間平均パワーを推定する第
１の平均パワー推定部、受話側損失挿入手段の出力信号
の時間平均パワーを推定する第２の平均パワー推定部、
受話側損失挿入手段の出力点から送話側損失挿入手段の
入力点の間の近端側帰還経路にて想定される最大遅延時
間における第２の平均パワー推定部の出力の最小値を求
める第１の最小値算出部、第１の平均パワー推定部の出
力値を第１の最小値算出部の出力値で除算する第１の除
算器を有して通話状態判定部により受話状態と判定され
たときにのみ各部の処理を更新して成り、遠端側帰還利
得推定部は、受話側損失挿入手段の入力信号の時間平均
パワーを推定する第３の平均パワー推定部、送話側損失
挿入手段の出力信号の時間平均パワーを推定する第４の
平均パワー推定部、送話側損失挿入手段の出力点から受
話側損失挿入手段の入力点の間の遠端側帰還経路にて想
定される最大遅延時間における第４の平均パワー推定部
の出力の最小値を求める第２の最小値算出部、第３の平
均パワー推定部の出力値を第２の最小値算出部の出力値
で除算する第２の除算部を有して通話状態判定部により
送話状態と判定されたときにのみ各部の処理を更新して
成ることを特徴とし、第１及び第２の最小値算出部にて
算出される最小値は帰還経路の伝達遅延時間の影響を低
減して高い精度で帰還利得を推定することができ、挿入
損失量を適切な値に調整して優れた同時通話性を確保す
ることができる。また、伝達遅延時間の変動に追従して
帰還利得を高い精度で推定することが可能である。しか
も、帰還利得の推定値が真の値よりも小さくなる場合は
少なく、総損失量の不足による信号経路の不安定化を防
ぐことができる。In order to achieve the above-mentioned object, according to the first aspect of the present invention, there is provided a microphone for outputting a collected sound as a voice signal on a transmitting side, and a microphone for amplifying a voice signal from the microphone. Amplifying means, a speaker that sounds in response to a sound signal on the receiving side, a second amplifying means for amplifying the sound signal output to the speaker, and inserting a predetermined amount of loss into a signal path on the transmitting side. Transmitting-side loss inserting means, receiving-side loss inserting means for inserting a predetermined amount of loss into the receiving-side signal path, and insertion-loss amount for adjusting insertion-loss amounts of the receiving-side loss inserting means and the transmitting-side loss inserting means. In a loudspeaker apparatus comprising an adjusting means, an insertion loss adjusting means monitors a voice signal of a transmitting side and a receiving side to determine a talking state and an input of a transmitting side loss inserting means. Signal and receiver insertion loss And a near-end feedback gain estimating unit for estimating the near-end feedback gain from the output signal of the receiver, and estimating the far-end feedback gain from the input signal of the receiving-side loss inserting unit and the output signal of the transmitting-side loss inserting unit. A far-end feedback gain estimator, a total loss calculator for calculating a total loss to be inserted into a closed loop based on the estimated values of the near-end feedback gain estimator and the far-end feedback gain estimator, An insertion loss amount distribution processing unit that determines each insertion loss amount of the transmission side loss insertion unit and the reception side insertion loss unit according to the determination result of the state determination unit and the calculated value of the total loss amount calculation unit; The end-side feedback gain estimating unit estimates a time average power of the input signal of the transmitting-side loss inserting means, and a second average power estimating unit estimates the time-average power of the output signal of the receiving-side loss inserting means. Average power estimator,
The minimum value of the output of the second average power estimator at the maximum delay time assumed in the near-end feedback path between the output point of the receiving-side loss inserting means and the input point of the transmitting-side loss inserting means is determined. 1 and a first divider for dividing the output value of the first average power estimating unit by the output value of the first minimum power calculating unit. The far-end feedback gain estimating unit estimates the time-average power of the input signal of the receiving-side loss insertion unit, and the transmission-side loss insertion A fourth average power estimator for estimating the time average power of the output signal of the means, assumed on the far end feedback path from the output point of the transmitting side loss inserting means to the input point of the receiving side loss inserting means. The minimum value of the output of the fourth average power estimator at the maximum delay time is And a second divider for dividing the output value of the third average power estimator by the output value of the second minimum calculator. The processing of each unit is updated only when it is determined that the state is determined. The minimum value calculated by the first and second minimum value calculation units reduces the influence of the transmission delay time of the feedback path. Thus, the feedback gain can be estimated with high accuracy, and the amount of insertion loss can be adjusted to an appropriate value to ensure excellent simultaneous communication. Further, it is possible to estimate the feedback gain with high accuracy by following the variation of the transmission delay time. In addition, the estimated value of the feedback gain is less likely to be smaller than the true value, and the instability of the signal path due to the shortage of the total loss can be prevented.
【００１０】上記目的を達成するために、請求項２の発
明は、集音した音を送話側の音声信号として出力するマ
イクロホンと、マイクロホンからの音声信号を増幅する
第１の増幅手段と、受話側の音声信号に応じて鳴動する
スピーカと、スピーカへ出力される音声信号を増幅する
第２の増幅手段と、送話側の信号経路に所定量の損失を
挿入する送話側損失挿入手段と、受話側の信号経路に所
定量の損失を挿入する受話側損失挿入手段と、受話側損
失挿入手段及び送話側損失挿入手段の挿入損失量を調整
する挿入損失量調整手段とを備えた拡声通話装置におい
て、挿入損失量調整手段は、送話側並びに受話側の音声
信号を監視して通話状態を判定する通話状態判定部と、
送話側損失挿入手段の入力信号並びに受話側損失挿入手
段の出力信号から近端側の帰還利得を推定する近端側帰
還利得推定部と、受話側損失挿入手段の入力信号並びに
送話側損失挿入手段の出力信号から遠端側の帰還利得を
推定する遠端側帰還利得推定部と、近端側帰還利得推定
部及び遠端側帰還利得推定部の各推定値に基づいて閉ル
ープに挿入すべき総損失量を算出する総損失量算出部
と、通話状態判定部の判定結果と総損失量算出部の算出
値に応じて送話側損失挿入手段及び受話側挿入損失手段
の各挿入損失量を決定する挿入損失量分配処理部とを具
備し、近端側帰還利得推定部は、送話側損失挿入手段の
入力信号の時間平均パワーを推定する第１の平均パワー
推定部、受話側損失挿入手段の出力信号の時間平均パワ
ーを推定する第２の平均パワー推定部、第２の平均パワ
ー推定部の出力を閉ループの最小伝達遅延時間以上の時
間だけ遅延させる第１の遅延部、第１の平均パワー推定
部の出力値を第１の遅延部の出力値で除算する第１の除
算器を有して通話状態判定部により受話状態と判定され
たときにのみ各部の処理を更新して成り、遠端側帰還利
得推定部は、受話側損失挿入手段の入力信号の時間平均
パワーを推定する第３の平均パワー推定部、送話側損失
挿入手段の出力信号の時間平均パワーを推定する第４の
平均パワー推定部、第４の平均パワー推定部の出力を閉
ループの最小伝達遅延時間以上の時間だけ遅延させる第
２の遅延部、第３の平均パワー推定部の出力値を第２の
遅延部の出力値で除算する第２の除算部を有して通話状
態判定部により送話状態と判定されたときにのみ各部の
処理を更新して成ることを特徴とし、第１及び第２の遅
延部の出力値は帰還経路の伝達遅延時間の影響を低減し
て高い精度で帰還利得を推定することができ、挿入損失
量を適切な値に調整して優れた同時通話性を確保するこ
とができる。また、帰還経路の伝達遅延時間が既知であ
って殆ど変動がないような場合に単純な演算処理により
帰還利得を推定することが可能となり、信号処理量の低
減化が図れる。According to another aspect of the present invention, there is provided a microphone for outputting a collected sound as an audio signal on a transmitting side, a first amplifying means for amplifying an audio signal from the microphone, A speaker that rings in response to a voice signal on the receiving side, a second amplifying unit that amplifies the voice signal output to the speaker, and a loss insertion unit on the transmitting side that inserts a predetermined amount of loss into a signal path on the transmitting side And a receiving-side loss inserting unit for inserting a predetermined amount of loss into a signal path on the receiving side, and an insertion-loss-amount adjusting unit for adjusting insertion loss amounts of the receiving-side loss-inserting unit and the transmitting-side loss-inserting unit. In the loudspeaker apparatus, the insertion loss adjusting means monitors a voice signal on the transmitting side and a voice signal on the receiving side to determine a communication state;
A near-end feedback gain estimator for estimating a near-end feedback gain from an input signal of the transmitting-side loss inserting means and an output signal of the receiving-side loss inserting means; an input signal of the receiving-side loss inserting means and a transmitting-side loss; A far-end feedback gain estimator for estimating the far-end feedback gain from the output signal of the insertion means, and a closed-loop feedback gain estimator and a closer-end feedback gain estimator are inserted into a closed loop based on the respective estimated values. A total loss amount calculation unit for calculating the total loss amount to be calculated, and each insertion loss amount of the transmission side loss insertion unit and the reception side insertion loss unit according to the determination result of the call state determination unit and the calculated value of the total loss amount calculation unit. And a near-end-side feedback gain estimator, wherein a first average power estimator for estimating the time average power of the input signal of the transmitting-side loss inserting means, a receiving-side loss A second method for estimating the time average power of the output signal of the insertion means; A first delay unit that delays the outputs of the average power estimator and the second average power estimator by a time equal to or longer than the minimum transmission delay time of the closed loop, and outputs the output values of the first average power estimator to the first delay unit It has a first divider for dividing by the output value, and updates the processing of each unit only when it is determined that the receiving state is determined by the call state determining unit. A third average power estimator for estimating the time average power of the input signal of the means, a fourth average power estimator for estimating the time average power of the output signal of the transmitting side loss insertion means, and a fourth average power estimator. And a second divider for dividing the output value of the third average power estimator by the output value of the second delay unit. Was determined to be in the transmission state by the call state determination unit The processing of each section is updated only when the output value of the first and second delay sections is reduced by the effect of the propagation delay time of the feedback path to estimate the feedback gain with high accuracy. By adjusting the insertion loss to an appropriate value, it is possible to ensure excellent simultaneous communication. In addition, when the transmission delay time of the feedback path is known and there is almost no change, it is possible to estimate the feedback gain by simple arithmetic processing, and the amount of signal processing can be reduced.
【００１１】請求項３の発明は、請求項１の発明におい
て、第１及び第２の除算器の出力値のうちから人間の音
声の音韻継続時間以上の所定時間内における最小値を算
出する第３及び第４の最小値算出部を近端側帰還利得推
定部並びに遠端側帰還利得推定部にそれぞれ設け、第３
及び第４の最小値算出部の出力値を近端側帰還利得推定
部並びに遠端側帰還利得推定部の推定値として成ること
を特徴とし、請求項１の発明の作用に加えて、所定時間
における帰還利得の推定値の最小値を用いて総損失量を
算出するため、所謂ダブルトーク状態において帰還利得
の推定値が真の値よりも大きくなるのを抑制することが
できる。According to a third aspect of the present invention, in the first aspect of the present invention, a minimum value within a predetermined period of time equal to or longer than a phoneme duration of a human voice is calculated from the output values of the first and second dividers. The third and fourth minimum value calculation units are provided in the near-end feedback gain estimation unit and the far-end feedback gain estimation unit, respectively.
And an output value of the fourth minimum value calculating section as an estimated value of the near-end feedback gain estimating section and the far-end feedback gain estimating section. Since the total loss is calculated using the minimum value of the estimated value of the feedback gain in the above, it is possible to suppress the estimated value of the feedback gain from becoming larger than the true value in a so-called double talk state.
【００１２】請求項４の発明は、請求項１の発明におい
て、適応フィルタを有し近端側帰還経路又は遠端側帰還
経路の少なくとも何れか一方に設けられるエコーキャン
セラと、適応フィルタの係数の収束値からエコーキャン
セラを設けた側の帰還経路の利得を近似的に算出する帰
還利得近似算出部と、この帰還利得近似算出部の出力値
とエコーキャンセラを設けた側の帰還利得推定部の帰還
利得推定値との比を求める第３の除算器と、第３の除算
器の出力値に応じてエコーキャンセラを設けた側の帰還
利得推定部の帰還利得推定値の最新の値と以前の値の何
れか一方を選択して総損失量算出部に出力する帰還利得
選択部とを備え、この帰還利得選択部は、第３の除算器
の出力値が所定の範囲内である場合には最新の帰還利得
推定値を選択し、第３の除算器の出力値が所定の範囲を
超えた場合には以前の帰還利得推定値を選択して成るこ
とを特徴とし、請求項１の発明の作用に加えて、帰還利
得近似算出部の近似的な推定値を参照して適切な推定値
を帰還利得選択部で選択しているから、通話路中に衝撃
性のノイズが混入した場合等の外乱により生じる推定誤
差を低減することができる。According to a fourth aspect of the present invention, in the first aspect of the present invention, there is provided an echo canceller which has an adaptive filter and is provided in at least one of the near-end feedback path and the far-end feedback path; A feedback gain approximation calculator for approximately calculating the gain of the feedback path on the side provided with the echo canceller from the convergence value, and an output value of the feedback gain approximation calculator and feedback from the feedback gain estimator on the side provided with the echo canceller A third divider for obtaining a ratio with the gain estimated value, and the latest value and the previous value of the feedback gain estimated value of the feedback gain estimator of the feedback gain estimator provided with the echo canceller according to the output value of the third divider And a feedback gain selector for selecting one of the two and outputting the selected value to the total loss calculator. The feedback gain selector is configured to output the latest value when the output value of the third divider is within a predetermined range. Choose the feedback gain estimate for 3. When the output value of the divider 3 exceeds a predetermined range, the previous feedback gain estimation value is selected, and in addition to the operation of the invention of claim 1, the feedback gain approximation calculation unit Since an appropriate estimated value is selected by the feedback gain selection unit with reference to the approximate estimated value, it is possible to reduce an estimation error caused by disturbance such as a case where impact noise is mixed in a communication path. .
【００１３】[0013]
【発明の実施の形態】（実施形態１）本実施形態の拡声
通話装置は、図１に示すよう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１と、マイクロホン
１からの音声信号を増幅する第１の増幅器２と、受話側
の音声信号に応じて鳴動するスピーカ３と、スピーカ３
へ出力される音声信号を増幅する第２の増幅器４と、送
話側の信号経路に所定量の損失を挿入する送話側損失挿
入手段たる送話信号減衰器５と、受話側の信号経路に所
定量の損失を挿入する受話側損失挿入手段たる受話信号
減衰器６と、送話信号減衰器５及び受話信号減衰器６の
損失量を調整する挿入損失量調整手段７とを備えてい
る。DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS (Embodiment 1) A loudspeaker apparatus according to the present embodiment has a microphone 1 for outputting a collected sound as a voice signal on the transmitting side as shown in FIG. 1, and a voice from the microphone 1. A first amplifier 2 for amplifying a signal, a speaker 3 that sounds in response to a voice signal on the receiving side, and a speaker 3
A second amplifier 4 for amplifying the audio signal output to the transmitting side, a transmitting side signal attenuator 5 as transmitting side loss inserting means for inserting a predetermined amount of loss into a transmitting side signal path, and a receiving side signal path. A receiving signal attenuator 6 serving as a receiving-side loss inserting means for inserting a predetermined amount of loss into the receiver, and an insertion loss adjusting means 7 for adjusting the loss of the transmitting signal attenuator 5 and the receiving signal attenuator 6. .
【００１４】また、挿入損失量調整手段７は、図２に示
すように送話側並びに受話側の音声信号を監視して通話
状態を判定する通話状態判定部８と、送話信号減衰器５
の入力信号ｓin並びに受話信号減衰器６の出力信号ｒ
outから近端側の帰還利得を推定する近端側帰還利得推
定部９と、受話信号減衰器６の入力信号ｒin並びに送話
信号減衰器５の出力信号ｓoutから遠端側の帰還利得を
推定する遠端側帰還利得推定部１０と、近端側帰還利得
推定部９及び遠端側帰還利得推定部１０の各推定値に基
づいて閉ループに挿入すべき総損失量を算出する総損失
量算出部１１と、通話状態判定部８の判定結果と総損失
量算出部１１の算出値に応じて送話信号減衰器５及び受
話信号減衰器６の各挿入損失量を決定する挿入損失量分
配処理部１２とを具備している。As shown in FIG. 2, the insertion loss adjusting means 7 monitors a voice signal on the transmitting side and a voice signal on the receiving side to determine a voice state, and a voice state attenuator 5.
Input signal s in and the output signal r of the reception signal attenuator 6
a near-end side feedback gain estimator 9 estimates the feedback gain of the near-end side from out, the input signal r in and transmission signal attenuator 5 the output signal s out from the far end of the feedback gain of the received signal attenuator 6 , And the total loss for calculating the total loss to be inserted into the closed loop based on the estimated values of the near-end feedback gain estimator 9 and the far-end feedback gain estimator 10. The amount of insertion loss for determining the amount of insertion loss of the transmission signal attenuator 5 and the amount of reception loss of the reception signal attenuator 6 in accordance with the determination result of the communication state determination unit 8 and the value calculated by the total loss amount calculation unit 11. And a distribution processing unit 12.
【００１５】さらに近端側帰還利得推定部９は、送話信
号減衰器５の入力信号ｓinの時間平均パワーを推定する
第１の平均パワー推定部１３と、受話信号減衰器６の出
力信号ｒoutの時間平均パワーを推定する第２の平均パ
ワー推定部１４と、受話信号減衰器６の出力点から送話
信号減衰器５の入力点の間の近端側帰還経路にて想定さ
れる最大遅延時間における第２の平均パワー推定部１４
の出力の最小値を求める第１の最小値算出部１５と、第
１の平均パワー推定部１３の出力値を第１の最小値算出
部１５の出力値で除算する第１の除算器１６とで構成さ
れる。また遠端側帰還利得推定部１０は、受話信号減衰
器６の入力信号ｒinの時間平均パワーを推定する第３の
平均パワー推定部１７と、送話信号減衰器５の出力信号
ｓoutの時間平均パワーを推定する第４の平均パワー推
定部１８と、送話信号減衰器５の出力点から受話信号減
衰器６の入力点の間の遠端側帰還経路にて想定される最
大遅延時間における第４の平均パワー推定部１８の出力
の最小値を求める第２の最小値算出部１９と、第３の平
均パワー推定部１７の出力値を第２の最小値算出部１９
の出力値で除算する第２の除算部２０とで構成される。
なお、挿入損失量調整手段７はＤＳＰ(Digital Signal
Processor)やＣＰＵ等を用いていて構成することがで
き、さらには近端側帰還利得推定部９並びに遠端側帰還
利得推定部１０のみをアナログ回路で構成してもよい。Further, the near-end feedback gain estimator 9 estimates a time average power of the input signal s in of the transmission signal attenuator 5 and an output signal of the reception signal attenuator 6. A second average power estimator 14 for estimating the time average power of r out and a near-end side feedback path between the output point of the received signal attenuator 6 and the input point of the transmitted signal attenuator 5 are assumed. Second average power estimator 14 at maximum delay time
A first minimum value calculating unit 15 for obtaining the minimum value of the output of the first and second power units; a first divider 16 for dividing the output value of the first average power estimating unit 13 by the output value of the first minimum value calculating unit 15; It consists of. The far end feedback gain estimator 10 includes a third average power estimation unit 17 for estimating the time average power of the input signal r in the received signal attenuator 6, the output signal s out of the transmission signal attenuator 5 A fourth average power estimator 18 for estimating time average power, and a maximum delay time assumed on a far-end side feedback path between an output point of the transmission signal attenuator 5 and an input point of the reception signal attenuator 6 , A second minimum value calculating unit 19 for obtaining the minimum value of the output of the fourth average power estimating unit 18, and an output value of the third average power estimating unit 17 being calculated by the second minimum value calculating unit 19.
And a second divider 20 for dividing by the output value of.
The insertion loss adjusting means 7 is provided with a DSP (Digital Signal).
Processor and CPU, etc., and only the near-end feedback gain estimator 9 and the far-end feedback gain estimator 10 may be configured by analog circuits.
【００１６】通話状態判定部８では送話信号減衰器５の
入出力信号ｓin，ｓout及び受話信号減衰器６の入出力
信号ｒin，ｒoutを監視し、これらの信号ｓin，ｓout，
ｒin，ｒoutのパワーレベルの大小関係並びに音声信号
の有無などの情報から通話状態が受話状態、送話状態、
遷移状態の何れであるかを判定し、その判定結果を挿入
損失量分配処理部１２に与えている。The call state judging unit 8 monitors the input / output signals s in and s out of the transmission signal attenuator 5 and the input / output signals r in and r out of the reception signal attenuator 6 and these signals s in and s. out ,
r in, r out power level of the magnitude relationship as well as the audio signal information from the call state is receiving state, such as the presence or absence of, the sending state,
It is determined which of the transition states it is in, and the result of the determination is given to the insertion loss distribution processor 12.
【００１７】近端側帰還利得推定部９を構成する第１の
平均パワー推定部１３は、整流平滑器や低域通過フィル
タ等を用いて送話信号減衰器５への入力信号ｓinの短時
間における時間平均パワーＬｓinを求めている。同じく
第２の平均パワー推定部１４は、受話信号減衰器６の出
力信号ｒoutの短時間における時間平均パワーＬｒoutを
求めている。また、第１の最小値算出部１５において
は、近端側帰還経路において考えられる最大の伝達遅延
時間の間、上記時間平均パワーＬｒoutを観測してその
最小値［Ｌｒout］minを求める。第１の除算器１６にお
いては、第１の平均パワー推定部１３の出力Ｌｓinと第
１の最小値算出部１５の出力［Ｌｒout］minとの比を算
出し、この値を近端側帰還利得の推定値として総損失量
算出部１１に出力する。The first average power estimator 13 constituting the near-end feedback gain estimator 9 uses a rectifier / smoothing device or a low-pass filter to shorten the input signal s in to the transmission signal attenuator 5. The time average power Ls in over time is determined. Similarly, the second average power estimating unit 14 obtains a short-time average power Lr out of the output signal r out of the reception signal attenuator 6. Further, the first minimum value calculating section 15 observes the time average power Lr out during the maximum transmission delay time conceivable in the near-end side feedback path, and obtains the minimum value [Lr out ] min. The first divider 16 calculates a ratio between the output Ls in of the first average power estimating unit 13 and the output [Lr out ] min of the first minimum value calculating unit 15, and calculates this value on the near end side. The feedback loss is output to the total loss calculator 11 as an estimated value.
【００１８】一方、遠端側帰還利得推定部１０を構成す
る第３の平均パワー推定部１７は、整流平滑器や低域通
過フィルタ等を用いて受話信号減衰器６への入力信号ｒ
inの短時間における時間平均パワーＬｒinを求めてい
る。同じく第４の平均パワー推定部１８は、送話信号減
衰器５の出力信号ｓoutの短時間における時間平均パワ
ーＬｓoutを求めている。また、第２の最小値算出部１
９においては、遠端側帰還経路において考えられる最大
遅延時間の間、上記時間平均パワーＬｓoutを観測して
その最小値［Ｌｓout］minを求める。第２の除算器２０
においては、第３の平均パワー推定部１７の出力Ｌｒin
と第２の最小値算出部１９の出力［Ｌｓou t］minとの比
を算出し、この値を遠端側帰還利得の推定値として総損
失量算出部１１に出力する。On the other hand, the third average power estimating unit 17 constituting the far-end feedback gain estimating unit 10 uses an input signal r to the reception signal attenuator 6 by using a rectifying smoother, a low-pass filter, or the like.
The time average power Lr in in a short time is obtained. Similarly, the fourth average power estimating unit 18 obtains the time average power Ls out of the output signal s out of the transmission signal attenuator 5 in a short time. Also, the second minimum value calculation unit 1
In step 9, the time average power Ls out is observed during the maximum delay time conceivable in the far-end feedback path, and the minimum value [Ls out ] min is obtained. Second divider 20
, The output Lr in of the third average power estimation unit 17
When the output to calculate the ratio of the [Ls ou t] min of the second minimum value calculation unit 19, and outputs the total loss amount calculation unit 11 this value as an estimate of the far-end feedback gain.
【００１９】ここで、近端側帰還利得推定部９における
上記推定処理は通話状態判定部８にて通話状態が受話状
態と判定されたときに更新され、遠端側帰還利得推定部
１０における上記推定処理は通話状態判定部８にて通話
状態が送話状態と判定されたときに更新される。Here, the above estimation processing in the near-end feedback gain estimating section 9 is updated when the talking state is determined to be in the receiving state by the talking state determining section 8, and the above-described processing in the far-end feedback gain estimating section 10 is performed. The estimation process is updated when the call state determination unit 8 determines that the call state is the transmission state.
【００２０】総損失量算出部１１においては、これらの
帰還利得推定値から所望の利得余裕（ハウリングマージ
ン）を得るために必要な総挿入損失量を算出し、その値
を挿入損失量分配処理部１２に出力する。そして、挿入
損失量分配処理部１２では、通話状態判定部８にて判定
された通話状態に応じた割合で総挿入損失量を送話信号
減衰器５と受話信号減衰器６に分配するように各減衰器
５，６の損失量を調整する。The total loss calculator 11 calculates the total insertion loss required to obtain a desired gain margin (howling margin) from these feedback gain estimation values, and uses that value as the insertion loss distribution processor. 12 is output. Then, the insertion loss amount distribution processing unit 12 distributes the total insertion loss amount to the transmission signal attenuator 5 and the reception signal attenuator 6 at a rate corresponding to the communication state determined by the communication state determination unit 8. The loss amounts of the attenuators 5 and 6 are adjusted.
【００２１】而して、本実施形態では第１及び第２の最
小値算出部１５，１９により近端側並びに遠端側の帰還
経路において考えられる最大の伝達遅延時間の間、第２
及び第４の平均パワー推定部１４，１８の出力値（時間
平均パワーＬｒout，Ｌｓout）を観測してその最小値
［Ｌｒout］min，［Ｌｓout］minを求め、その値を用い
て帰還利得の推定値を算出しているから、帰還経路の伝
達遅延時間の影響を低減して帰還利得の推定精度を向上
することができるとともに、上記伝達遅延時間が変動し
てもそれに追従して高い精度で帰還利得を推定すること
ができる。また、本実施形態では帰還利得の推定値が真
の値よりも小さくなる場合は少なく、総損失量の不足に
よる信号経路の不安定化を防ぐことができるという利点
もある。In the present embodiment, the first and second minimum value calculators 15 and 19 perform the second transmission during the maximum transmission delay time conceivable in the near-end and far-end return paths.
And the output values (time-average power Lr out , Ls out ) of the fourth average power estimating units 14 and 18 are observed, and their minimum values [Lr out ] min and [Ls out ] min are obtained. Since the estimated value of the feedback gain is calculated, the effect of the propagation delay time of the feedback path can be reduced to improve the estimation accuracy of the feedback gain. The feedback gain can be estimated with high accuracy. Further, in the present embodiment, the estimated value of the feedback gain is less likely to be smaller than the true value, and there is an advantage that the instability of the signal path due to the shortage of the total loss can be prevented.
【００２２】（実施形態２）本発明の実施形態２におけ
る挿入損失量調整手段７のブロック図を図３に示す。本
実施形態は、実施形態１の挿入損失量調整手段７におけ
る第１及び第２の最小値算出部１５，１９を、それぞれ
第１の遅延器２１及び第２の遅延器２２に置き換えたも
のであり、その他の各部の構成及び動作は実施形態１と
共通するので、共通する部分には同一の符号を付して説
明を省略する。(Embodiment 2) FIG. 3 shows a block diagram of the insertion loss adjusting means 7 in Embodiment 2 of the present invention. In the present embodiment, the first and second minimum value calculators 15 and 19 in the insertion loss adjusting means 7 of the first embodiment are replaced with a first delay unit 21 and a second delay unit 22, respectively. The configuration and operation of the other parts are common to those of the first embodiment. Therefore, the common parts are denoted by the same reference numerals and description thereof is omitted.
【００２３】第１及び第２の遅延器２１，２２は、それ
ぞれ近端側の帰還経路並びに遠端側の帰還経路の構成要
素や諸条件（マイクロホン１−スピーカ３間の距離な
ど）により一意に定まる最短の伝達遅延時間以上の一定
時間だけ第２の平均パワー推定部１４及び第４の平均パ
ワー推定部１８の出力信号を遅延させて第１及び第２の
除算器１６，２０に出力するものである。すなわち、こ
れら近端側及び遠端側の帰還経路の伝達遅延時間を各々
ＴN，ＴFとすると、時刻ｔにおける近端側の帰還利得推
定値はＬｓin(t)／Ｌｒout（t-ＴN)、遠端側の帰還利得
推定値はＬｒin(t)／Ｌｓout（t-ＴF)としてそれぞれ表
される。The first and second delay units 21 and 22 are uniquely determined by the components and conditions (distance between the microphone 1 and the speaker 3) of the near-end return path and the far-end return path, respectively. The output signals of the second average power estimating unit 14 and the fourth average power estimating unit 18 are delayed by a fixed time equal to or longer than the determined shortest transmission delay time and output to the first and second dividers 16 and 20. It is. That is, assuming that the transmission delay times of the near-end and far-end feedback paths are T N and T F , respectively, the estimated value of the near-end feedback gain at time t is Ls in (t) / Lr out (t− T N ) and the feedback gain estimation value at the far end are expressed as Lr in (t) / Ls out (t-T F ), respectively.
【００２４】而して本実施形態によれば、帰還経路の伝
達遅延時間が既知であり且つ変動が少ない場合に第１及
び第２の遅延器２１，２２で上記伝達遅延時間以上の一
定時間だけ第２の平均パワー推定部１４及び第４の平均
パワー推定部１８の出力信号を遅延させることにより、
伝達遅延時間を考慮して精度よく帰還利得を推定するこ
とができる。しかも、第１及び第２の遅延器２１，２２
は実施形態１における第１及び第２の最小値算出部１
５，１９に比較して構成が簡単であり、コストダウンが
図れるという利点がある。According to this embodiment, when the transmission delay time of the feedback path is known and the fluctuation is small, the first and second delay units 21 and 22 use the transmission delay time for a fixed time longer than the transmission delay time. By delaying the output signals of the second average power estimator 14 and the fourth average power estimator 18,
The feedback gain can be accurately estimated in consideration of the propagation delay time. Moreover, the first and second delay units 21 and 22
Is the first and second minimum value calculation units 1 in the first embodiment.
There is an advantage that the configuration is simpler than that of Nos. 5 and 19, and the cost can be reduced.
【００２５】（実施形態３）本発明の実施形態３におけ
る挿入損失量調整手段７のブロック図を図４に示す。本
実施形態は、実施形態１における挿入損失量調整手段７
に対して、第１及び第２の除算器１６，２０の出力側に
それぞれ第３及び第４の最小値算出部２３，２４を設
け、これら第３及び第４の最小値算出部２３，２４の出
力を近端側及び遠端側の帰還利得推定値として総損失量
算出部１１に入力するものであり、その他の各部の構成
及び動作は実施形態１と共通するので、共通する部分に
は同一の符号を付して説明を省略する。(Embodiment 3) FIG. 4 is a block diagram of the insertion loss adjusting means 7 according to Embodiment 3 of the present invention. In this embodiment, the insertion loss adjusting means 7 in the first embodiment is used.
, The third and fourth minimum value calculation units 23 and 24 are provided on the output sides of the first and second dividers 16 and 20, respectively, and these third and fourth minimum value calculation units 23 and 24 are provided. Are input to the total loss calculating unit 11 as feedback gain estimation values on the near end and the far end, and the configuration and operation of the other units are the same as those in the first embodiment. The same reference numerals are given and the description is omitted.
【００２６】第３及び第４の最小値算出部２３，２４
は、ＲＡＭ等の記憶手段を用いて人間の音声の音韻継続
時間以上の一定時間だけ第１及び第２の除算器１６，２
０の出力値を保持し、その保持した値の最小値をその時
点での帰還利得推定値として抽出し総損失量算出部１１
に出力する。このような処理により、送話信号減衰器５
の入力信号ｓin並びに受話信号減衰器６の入力信号ｒin
にエコー成分（受話信号減衰器６の出力信号ｒout及び
送話信号減衰器５の出力信号ｓoutが各々近端側及び遠
端側の帰還経路を経て回り込む信号成分）以外の成分、
すなわち近端側の話者の音声信号、遠端側の話者の音声
信号並びに周囲騒音が含まれる場合にそれらの影響が最
小となる瞬間に推定された帰還利得を得ることができ、
推定精度の向上が図れるのである。Third and fourth minimum value calculators 23 and 24
Are stored in the first and second dividers 16 and 2 for a certain period of time equal to or longer than the duration of the phoneme of a human voice using a storage unit such as a RAM.
0, and the minimum value of the held values is extracted as the feedback gain estimation value at that time, and the total loss amount calculation unit 11
Output to By such processing, the transmission signal attenuator 5
Input signal s in as well as the input signal r in the received signal attenuator 6
Components other than echo components (signal components in which the output signal r out of the reception signal attenuator 6 and the output signal s out of the transmission signal attenuator 5 wrap around via the near-end and far-end feedback paths, respectively)
That is, when the voice signal of the near-end speaker, the voice signal of the far-end speaker and the ambient noise are included, the feedback gain estimated at the moment when their influence is minimized can be obtained.
The accuracy of the estimation can be improved.
【００２７】上述のように本実施形態によれば、近端側
と遠端側の話者が略同時に話す状態、所謂ダブルトーク
状態において帰還利得の推定値が真の値よりも大きくな
るのを抑制することができる。As described above, according to the present embodiment, it is assumed that the feedback gain estimated value becomes larger than the true value in a state where the near-end and far-end speakers speak almost simultaneously, that is, a so-called double talk state. Can be suppressed.
【００２８】（実施形態４）本実施形態の拡声通話装置
は、図５に示すように送話側の信号経路に所定量の損失
を挿入する送話信号減衰器５と、受話側の信号経路に所
定量の損失を挿入する受話信号減衰器６と、送話信号減
衰器５及び受話信号減衰器６の損失量を調整する挿入損
失量調整手段７と、適応フィルタ２５ａを有し近端側帰
還経路に設けられるエコーキャンセラ２５と、適応フィ
ルタ２５ａの係数の収束値からエコーキャンセラ２５を
設けた近端側の帰還経路の利得を近似的に算出する帰還
利得近似算出部２６と、この帰還利得近似算出部２６の
出力値と近端側帰還利得推定部９の帰還利得推定値との
比を求める第３の除算器２７と、第３の除算器２７の出
力値に応じて近端側帰還利得推定部９の帰還利得推定値
の最新の値と以前の値の何れか一方を選択して総損失量
算出部１１に出力する帰還利得選択部２８とを備えてい
る。なお、図示はしていないが実施形態１と同様にマイ
クロホン１、第１の増幅器２、スピーカ３並びに第２の
増幅器４を備えていることは言うまでもない。また、実
施形態１と共通する構成については同一の符号を付して
説明を省略する。(Embodiment 4) As shown in FIG. 5, a loudspeaker apparatus according to this embodiment comprises a transmission signal attenuator 5 for inserting a predetermined amount of loss into a transmission signal path, and a reception signal path. A receiving signal attenuator 6 for inserting a predetermined amount of loss, insertion loss amount adjusting means 7 for adjusting the loss amounts of the transmitting signal attenuator 5 and the receiving signal attenuator 6, and an near end side having an adaptive filter 25a. An echo canceller 25 provided on the feedback path, a feedback gain approximation calculator 26 for approximately calculating the gain of the feedback path on the near end side provided with the echo canceller 25 from the convergence value of the coefficient of the adaptive filter 25a, A third divider 27 for obtaining a ratio between the output value of the approximation calculating section 26 and the feedback gain estimated value of the near-end feedback gain estimating section 9, and the near-end feedback section according to the output value of the third divider 27. The latest and previous values of the feedback gain estimation value of the gain estimation unit 9 And a feedback gain selection unit 28 for selecting and outputting one of the values in the total loss amount calculation unit 11. Although not shown, it goes without saying that the microphone 1, the first amplifier 2, the speaker 3, and the second amplifier 4 are provided as in the first embodiment. The same components as those in the first embodiment are denoted by the same reference numerals, and description thereof is omitted.
【００２９】エコーキャンセラ２５は、従来周知のよう
に近端側帰還経路のインパルス応答を適応フィルタ２５
ａにより同定し、参照信号（受話信号減衰器６の出力信
号ｒ out）からエコー信号を推定して減算器により相殺
して近端側において生じるエコーを消去するものであ
る。適応フィルタ２５ａの係数が収束した状態では、適
応フィルタ２５ａの係数は近端側の帰還経路のインパル
ス応答を近似しているので、その係数値から帰還利得を
算出することができる。帰還利得近似算出部２６はエコ
ーキャンセラ２５における適応フィルタ２５ａの係数か
ら近端側帰還利得を算出する。すなわち、適応フィルタ
２５ａの係数列をｈ0，ｈ1，ｈ2，…，ｈI -1（Ｉは適応
フィルタ２５ａのタップ数）とすると、その２乗和（＝
ｈ0 2＋ｈ1 2＋ｈ2 2＋…＋ｈI-1 2）から帰還利得を近似的
に算出することができる。The echo canceller 25 is, as conventionally known,
The impulse response of the near end feedback path to the adaptive filter 25
a, and the reference signal (the output signal of the reception signal attenuator 6)
Number r out) To estimate the echo signal and cancel it out by the subtractor
To eliminate the echo generated at the near end
You. When the coefficients of the adaptive filter 25a converge,
The coefficient of the response filter 25a is the impulse of the return path on the near end side.
Response gain, the feedback gain is calculated from the coefficient value.
Can be calculated. The feedback gain approximation calculator 26 is eco-friendly.
-Coefficient of adaptive filter 25a in canceller 25?
Then, the near-end feedback gain is calculated. That is, the adaptive filter
The coefficient sequence of 25a is h0, H1, HTwo, ..., hI -1(I is adaptive
Assuming the number of taps of the filter 25a), the sum of squares (=
h0 Two+ H1 Two+ HTwo Two+ ... + hI-1 Two) Approximate feedback gain
Can be calculated.
【００３０】第３の除算器２７は、帰還利得近似算出部
２６の出力信号と近端側帰還利得推定部９の出力信号と
の比を求める。帰還利得選択部２８は、第３の除算器２
７で求めた上記比の値が所定の範囲内か否かを判断し、
上記比の値が所定範囲内であれば近端側帰還利得推定部
９の出力信号を選択して総損失量算出部１１に出力し、
上記比の値が所定範囲外であればそれ以前の時刻におい
て所定範囲内と判断された上記比の値を選択して総損失
量算出部１１に出力する。すなわち、上記比の値が１に
近い場合は、近端側帰還利得推定部９の推定値には誤差
が少ないと考えられるが、上記比の値が１と離れている
場合には誤差が多く含まれている可能性が高いから、上
記所定範囲を１を中心とした許容誤差範囲とし、許容誤
差範囲は挿入損失量が所望の利得余裕（ハウリングマー
ジン）を得ることができるように決定される。The third divider 27 calculates the ratio between the output signal of the feedback gain approximation calculator 26 and the output signal of the near-end feedback gain estimator 9. The feedback gain selection unit 28 includes the third divider 2
It is determined whether the value of the ratio obtained in step 7 is within a predetermined range,
If the value of the ratio is within a predetermined range, the output signal of the near-end side feedback gain estimator 9 is selected and output to the total loss calculator 11,
If the value of the ratio is outside the predetermined range, the value of the ratio determined to be within the predetermined range at a time before that is selected and output to the total loss amount calculation unit 11. That is, when the value of the above ratio is close to 1, it is considered that the error is small in the estimated value of the near-end side feedback gain estimating unit 9, but when the value of the above ratio is far from 1, the error is large. Since there is a high possibility of being included, the predetermined range is set as an allowable error range centered at 1, and the allowable error range is determined so that the insertion loss can obtain a desired gain margin (howling margin). .
【００３１】上述のように本実施形態によれば、エコー
キャンセラ２５の適応フィルタ係数から帰還利得近似算
出部２６にて算出した帰還利得の推定値を参照して近端
側帰還利得推定部９における帰還利得推定値の妥当性を
確認するため、通話路中に衝撃性のノイズが混入した場
合等の外乱により生じる推定誤差を低減することができ
る。なお、本実施形態ではエコーキャンセラ２５を近端
側の帰還経路のみに設けたが、遠端側の帰還経路のみあ
るいは近端側と遠端側の両方の帰還経路にエコーキャン
セラ２５を設けてもよい。なお、本実施形態が効果を発
揮するのは、エコーキャンセラによるエコー消去量が所
定のしきい値を越えた場合のみに限る。As described above, according to this embodiment, the near-end feedback gain estimating unit 9 refers to the feedback gain estimation value calculated by the feedback gain approximation calculating unit 26 from the adaptive filter coefficient of the echo canceller 25. In order to confirm the validity of the feedback gain estimation value, it is possible to reduce an estimation error caused by disturbance such as a case where impact noise is mixed in a communication path. In the present embodiment, the echo canceller 25 is provided only on the return path on the near end side. However, the echo canceller 25 may be provided only on the return path on the far end side, or on both the return paths on the near and far end sides. Good. The present embodiment is effective only when the amount of echo cancellation by the echo canceller exceeds a predetermined threshold.
【００３２】[0032]
【発明の効果】請求項１の発明は、集音した音を送話側
の音声信号として出力するマイクロホンと、マイクロホ
ンからの音声信号を増幅する第１の増幅手段と、受話側
の音声信号に応じて鳴動するスピーカと、スピーカへ出
力される音声信号を増幅する第２の増幅手段と、送話側
の信号経路に所定量の損失を挿入する送話側損失挿入手
段と、受話側の信号経路に所定量の損失を挿入する受話
側損失挿入手段と、受話側損失挿入手段及び送話側損失
挿入手段の挿入損失量を調整する挿入損失量調整手段と
を備えた拡声通話装置において、挿入損失量調整手段
は、送話側並びに受話側の音声信号を監視して通話状態
を判定する通話状態判定部と、送話側損失挿入手段の入
力信号並びに受話側損失挿入手段の出力信号から近端側
の帰還利得を推定する近端側帰還利得推定部と、受話側
損失挿入手段の入力信号並びに送話側損失挿入手段の出
力信号から遠端側の帰還利得を推定する遠端側帰還利得
推定部と、近端側帰還利得推定部及び遠端側帰還利得推
定部の各推定値に基づいて閉ループに挿入すべき総損失
量を算出する総損失量算出部と、通話状態判定部の判定
結果と総損失量算出部の算出値に応じて送話側損失挿入
手段及び受話側挿入損失手段の各挿入損失量を決定する
挿入損失量分配処理部とを具備し、近端側帰還利得推定
部は、送話側損失挿入手段の入力信号の時間平均パワー
を推定する第１の平均パワー推定部、受話側損失挿入手
段の出力信号の時間平均パワーを推定する第２の平均パ
ワー推定部、受話側損失挿入手段の出力点から送話側損
失挿入手段の入力点の間の近端側帰還経路にて想定され
る最大遅延時間における第２の平均パワー推定部の出力
の最小値を求める第１の最小値算出部、第１の平均パワ
ー推定部の出力値を第１の最小値算出部の出力値で除算
する第１の除算器を有して通話状態判定部により受話状
態と判定されたときにのみ各部の処理を更新して成り、
遠端側帰還利得推定部は、受話側損失挿入手段の入力信
号の時間平均パワーを推定する第３の平均パワー推定
部、送話側損失挿入手段の出力信号の時間平均パワーを
推定する第４の平均パワー推定部、送話側損失挿入手段
の出力点から受話側損失挿入手段の入力点の間の遠端側
帰還経路にて想定される最大遅延時間における第４の平
均パワー推定部の出力の最小値を求める第２の最小値算
出部、第３の平均パワー推定部の出力値を第２の最小値
算出部の出力値で除算する第２の除算部を有して通話状
態判定部により送話状態と判定されたときにのみ各部の
処理を更新して成るので、第１及び第２の最小値算出部
にて算出される最小値は帰還経路の伝達遅延時間の影響
を低減して高い精度で帰還利得を推定することができ、
挿入損失量を適切な値に調整して優れた同時通話性を確
保することができるという効果がある。また、伝達遅延
時間の変動に追従して帰還利得を高い精度で推定するこ
とが可能であり、しかも、帰還利得の推定値が真の値よ
りも小さくなる場合は少なく、総損失量の不足による信
号経路の不安定化を防ぐことができるという効果があ
る。According to the first aspect of the present invention, there is provided a microphone for outputting a collected sound as an audio signal on a transmitting side, a first amplifying means for amplifying an audio signal from the microphone, and an audio signal on a receiving side. A speaker that sounds in response to the signal, second amplifying means for amplifying an audio signal output to the speaker, transmitting-side loss insertion means for inserting a predetermined amount of loss into a transmitting-side signal path, and a receiving-side signal. A loudspeaker apparatus comprising: a receiver-side loss insertion unit that inserts a predetermined amount of loss into a path; and an insertion-loss-amount adjusting unit that adjusts an insertion loss amount of the receiver-side loss insertion unit and the transmission-side loss insertion unit. The loss amount adjusting means monitors a voice signal on the transmitting side and the receiving side to determine a talking state, and a call state determining section, and an input signal of the transmitting side loss inserting section and an output signal of the receiving side loss inserting section, Estimate the feedback gain at the end A near-end feedback gain estimating unit; a far-end feedback gain estimating unit for estimating a far-end feedback gain from an input signal of the receiving-side loss inserting unit and an output signal of the transmitting-side loss inserting unit; A total loss amount calculation unit that calculates a total loss amount to be inserted into the closed loop based on the estimated values of the gain estimation unit and the far-end feedback gain estimation unit; and a determination result of the call state determination unit and a total loss amount calculation unit. An insertion loss amount distribution processing unit that determines each insertion loss amount of the transmission-side loss insertion unit and the reception-side insertion loss unit according to the calculated value, and the near-end feedback gain estimation unit includes a transmission-side loss insertion unit. A first average power estimator for estimating the time average power of the input signal of the means, a second average power estimator for estimating the time average power of the output signal of the receiver loss insertion means, and an output point of the receiver loss insertion means. -End feedback between the input point of the loss insertion means and the transmitter A first minimum value calculator for obtaining the minimum value of the output of the second average power estimator at the maximum delay time assumed on the road, and a first minimum value calculator for calculating the output value of the first average power estimator It has a first divider that divides by the output value of and updates the processing of each unit only when it is determined that the telephone is in the receiving state by the call state determining unit.
The far-end feedback gain estimating unit estimates a time-average power of the input signal of the receiving-side loss insertion unit, and a fourth average-power estimation unit estimates the time-average power of the output signal of the transmitting-side loss insertion unit. The average power estimating section, the output of the fourth average power estimating section at the maximum delay time assumed in the far end feedback path from the output point of the transmitting side loss inserting means to the input point of the receiving side loss inserting means. A second minimum value calculating unit for obtaining the minimum value of the second average value and a second dividing unit for dividing an output value of the third average power estimating unit by an output value of the second minimum value calculating unit, and a call state determining unit. , The processing of each unit is updated only when it is determined to be in the transmission state, so that the minimum value calculated by the first and second minimum value calculation units reduces the influence of the transmission delay time of the feedback path. Feedback gain with high accuracy,
By adjusting the insertion loss to an appropriate value, there is an effect that excellent simultaneous callability can be secured. In addition, it is possible to estimate the feedback gain with high accuracy by following the variation of the propagation delay time, and furthermore, the estimated value of the feedback gain is rarely smaller than the true value, which is small. There is an effect that instability of the signal path can be prevented.
【００３３】請求項２の発明は、集音した音を送話側の
音声信号として出力するマイクロホンと、マイクロホン
からの音声信号を増幅する第１の増幅手段と、受話側の
音声信号に応じて鳴動するスピーカと、スピーカへ出力
される音声信号を増幅する第２の増幅手段と、送話側の
信号経路に所定量の損失を挿入する送話側損失挿入手段
と、受話側の信号経路に所定量の損失を挿入する受話側
損失挿入手段と、受話側損失挿入手段及び送話側損失挿
入手段の挿入損失量を調整する挿入損失量調整手段とを
備えた拡声通話装置において、挿入損失量調整手段は、
送話側並びに受話側の音声信号を監視して通話状態を判
定する通話状態判定部と、送話側損失挿入手段の入力信
号並びに受話側損失挿入手段の出力信号から近端側の帰
還利得を推定する近端側帰還利得推定部と、受話側損失
挿入手段の入力信号並びに送話側損失挿入手段の出力信
号から遠端側の帰還利得を推定する遠端側帰還利得推定
部と、近端側帰還利得推定部及び遠端側帰還利得推定部
の各推定値に基づいて閉ループに挿入すべき総損失量を
算出する総損失量算出部と、通話状態判定部の判定結果
と総損失量算出部の算出値に応じて送話側損失挿入手段
及び受話側挿入損失手段の各挿入損失量を決定する挿入
損失量分配処理部とを具備し、近端側帰還利得推定部
は、送話側損失挿入手段の入力信号の時間平均パワーを
推定する第１の平均パワー推定部、受話側損失挿入手段
の出力信号の時間平均パワーを推定する第２の平均パワ
ー推定部、第２の平均パワー推定部の出力を閉ループの
最小伝達遅延時間以上の時間だけ遅延させる第１の遅延
部、第１の平均パワー推定部の出力値を第１の遅延部の
出力値で除算する第１の除算器を有して通話状態判定部
により受話状態と判定されたときにのみ各部の処理を更
新して成り、遠端側帰還利得推定部は、受話側損失挿入
手段の入力信号の時間平均パワーを推定する第３の平均
パワー推定部、送話側損失挿入手段の出力信号の時間平
均パワーを推定する第４の平均パワー推定部、第４の平
均パワー推定部の出力を閉ループの最小伝達遅延時間以
上の時間だけ遅延させる第２の遅延部、第３の平均パワ
ー推定部の出力値を第２の遅延部の出力値で除算する第
２の除算部を有して通話状態判定部により送話状態と判
定されたときにのみ各部の処理を更新して成るので、第
１及び第２の遅延部の出力値は帰還経路の伝達遅延時間
の影響を低減して高い精度で帰還利得を推定することが
でき、挿入損失量を適切な値に調整して優れた同時通話
性を確保することができるという効果がある。また、帰
還経路の伝達遅延時間が既知であって殆ど変動がないよ
うな場合に単純な演算処理により帰還利得を推定するこ
とが可能となり、信号処理量の低減化が図れるという効
果がある。According to a second aspect of the present invention, there is provided a microphone for outputting a collected sound as an audio signal on the transmitting side, a first amplifying means for amplifying the audio signal from the microphone, and a function of the audio signal on the receiving side. A ringing speaker, a second amplifying unit for amplifying an audio signal output to the speaker, a transmitting-side loss inserting unit for inserting a predetermined amount of loss into a transmitting-side signal path, and a A loudspeaker apparatus comprising: a receiving-side loss inserting unit for inserting a predetermined amount of loss; and an insertion-loss-amount adjusting unit for adjusting an insertion loss amount of the receiving-side loss-inserting unit and the transmitting-side loss-inserting unit. The adjusting means is
A call state determination unit that monitors a voice signal on the transmitting side and a receiving side to determine a communication state; and a feedback gain on the near end side based on an input signal of the transmitting side loss inserting unit and an output signal of the receiving side loss inserting unit. A near-end feedback gain estimating unit for estimating, a far-end feedback gain estimating unit for estimating a far-end feedback gain from an input signal of the receiving-side loss inserting unit and an output signal of the transmitting-side loss inserting unit; A total loss calculator for calculating the total loss to be inserted into the closed loop based on the estimated values of the side feedback gain estimator and the far end feedback gain estimator; a determination result of the call state determination unit and a total loss calculation The insertion loss amount distribution processing unit that determines each insertion loss amount of the transmission-side loss insertion unit and the reception-side insertion loss unit according to the calculated value of the unit. First average for estimating the time average power of the input signal of the loss insertion means A second average power estimator for estimating the time average power of the output signal of the receiver-side loss insertion means; and a second average power estimator for delaying the output of the second average power estimator by a time equal to or longer than the minimum transmission delay time of the closed loop. 1 delay unit, a first divider for dividing the output value of the first average power estimating unit by the output value of the first delay unit, and only when the call state determination unit determines that the telephone is in the receiving state. The far-end feedback gain estimating unit estimates the time average power of the input signal of the receiving-side loss insertion unit, and the output signal of the transmitting-side loss insertion unit. A fourth average power estimator for estimating the time average power of the second average power estimator, a second delay unit for delaying the output of the fourth average power estimator by a time equal to or longer than the minimum transmission delay time of the closed loop, and a third average power estimator Output value of the second delay unit. Since the processing of each unit is updated only when it is determined that the communication state is determined by the call state determination unit by the second division unit that divides by the value, the output values of the first and second delay units are It is possible to estimate the feedback gain with high accuracy by reducing the influence of the propagation delay time of the feedback path, and it is possible to adjust the amount of insertion loss to an appropriate value to ensure excellent simultaneous communication. . In addition, when the transmission delay time of the feedback path is known and there is almost no change, it is possible to estimate the feedback gain by simple arithmetic processing, and there is an effect that the amount of signal processing can be reduced.
【００３４】請求項３の発明は、第１及び第２の除算器
の出力値のうちから人間の音声の音韻継続時間以上の所
定時間内における最小値を算出する第３及び第４の最小
値算出部を近端側帰還利得推定部並びに遠端側帰還利得
推定部にそれぞれ設け、第３及び第４の最小値算出部の
出力値を近端側帰還利得推定部並びに遠端側帰還利得推
定部の推定値として成るので、請求項１の発明の効果に
加えて、所定時間における帰還利得の推定値の最小値を
用いて総損失量を算出するため、所謂ダブルトーク状態
において帰還利得の推定値が真の値よりも大きくなるの
を抑制することができるという効果がある。According to a third aspect of the present invention, a third and a fourth minimum value for calculating a minimum value within a predetermined time equal to or longer than a phoneme duration of a human voice from the output values of the first and second dividers. The calculation unit is provided in each of the near-end feedback gain estimation unit and the far-end feedback gain estimation unit, and the output values of the third and fourth minimum value calculation units are used as the near-end feedback gain estimation unit and the far-end feedback gain estimation. Since the total loss is calculated using the minimum value of the estimated value of the feedback gain in a predetermined time, in addition to the effect of the invention of claim 1, the feedback gain is estimated in a so-called double talk state. There is an effect that the value can be suppressed from becoming larger than the true value.
【００３５】請求項４の発明は、適応フィルタを有し近
端側帰還経路又は遠端側帰還経路の少なくとも何れか一
方に設けられるエコーキャンセラと、適応フィルタの係
数の収束値からエコーキャンセラを設けた側の帰還経路
の利得を近似的に算出する帰還利得近似算出部と、この
帰還利得近似算出部の出力値とエコーキャンセラを設け
た側の帰還利得推定部の帰還利得推定値との比を求める
第３の除算器と、第３の除算器の出力値に応じてエコー
キャンセラを設けた側の帰還利得推定部の帰還利得推定
値の最新の値と以前の値の何れか一方を選択して総損失
量算出部に出力する帰還利得選択部とを備え、この帰還
利得選択部は、第３の除算器の出力値が所定の範囲内で
ある場合には最新の帰還利得推定値を選択し、第３の除
算器の出力値が所定の範囲を超えた場合には以前の帰還
利得推定値を選択して成るので、請求項１の発明の効果
に加えて、帰還利得近似算出部の近似的な推定値を参照
して適切な推定値を帰還利得選択部で選択しているか
ら、通話路中に衝撃性のノイズが混入した場合等の外乱
により生じる推定誤差を低減することができるという効
果がある。According to a fourth aspect of the present invention, there is provided an echo canceller which has an adaptive filter and is provided on at least one of the near-end feedback path and the far-end feedback path, and an echo canceller based on a convergence value of coefficients of the adaptive filter. And a feedback gain approximation calculation unit for approximately calculating the gain of the feedback path on the side of the feedback path, and a ratio between the output value of the feedback gain approximation calculation unit and the feedback gain estimation value of the feedback gain estimation unit on the side provided with the echo canceller. A third divider to be obtained, and either the latest value or the previous value of the feedback gain estimation value of the feedback gain estimator of the feedback gain estimator provided with the echo canceller is selected according to the output value of the third divider. And a feedback gain selector that outputs the total feedback loss to the total loss calculator. The feedback gain selector selects the latest feedback gain estimation value when the output value of the third divider is within a predetermined range. And the output value of the third divider is If the value exceeds the range, the previous feedback gain estimation value is selected, so that in addition to the effect of the first aspect of the present invention, an appropriate estimation is performed by referring to the approximate estimation value of the feedback gain approximation calculation unit. Since the value is selected by the feedback gain selection unit, there is an effect that an estimation error caused by a disturbance such as a case where impact noise is mixed in a communication path can be reduced.
【図１】実施形態１の全体構成を示すブロック図であ
る。FIG. 1 is a block diagram illustrating an overall configuration of a first embodiment.
【図２】同上における挿入損失量調整手段を示すブロッ
ク図である。FIG. 2 is a block diagram showing an insertion loss amount adjusting means in the same as above.
【図３】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。FIG. 3 is a block diagram illustrating an insertion loss adjusting unit according to a second embodiment.
【図４】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。FIG. 4 is a block diagram illustrating an insertion loss adjusting unit according to a third embodiment.
【図５】実施形態４の一部省略した全体構成を示すブロ
ック図である。FIG. 5 is a block diagram showing an entire configuration of a fourth embodiment with a part omitted;
【図６】従来例を示すブロック図である。FIG. 6 is a block diagram showing a conventional example.
１ マイクロホン ２ 第１の増幅器 ３ スピーカ ４ 第２の増幅器 ５ 送話信号減衰器 ６ 受話信号減衰器 ７ 挿入損失量調整手段 Reference Signs List 1 microphone 2 first amplifier 3 speaker 4 second amplifier 5 transmission signal attenuator 6 reception signal attenuator 7 insertion loss adjusting means
───────────────────────────────────────────────────── フロントページの続き (72)発明者 寺澤 章 大阪府門真市大字門真1048番地松下電工株 式会社内 Ｆターム(参考） 5K027 BB03 DD07 DD10 HH01 5K038 CC02 FF10 FF13 5K046 AA01 BB01 CC30 DD21 HH37 HH58 HH77 HH78 ────────────────────────────────────────────────── ─── Continuing on the front page (72) Inventor Akira Terasawa 1048 Kazuma Kadoma, Osaka Pref. Matsushita Electric Works Co., Ltd.F-term (reference) HH78
Claims (4)
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、送話側の信号経路に所定量
の損失を挿入する送話側損失挿入手段と、受話側の信号
経路に所定量の損失を挿入する受話側損失挿入手段と、
受話側損失挿入手段及び送話側損失挿入手段の挿入損失
量を調整する挿入損失量調整手段とを備えた拡声通話装
置において、挿入損失量調整手段は、送話側並びに受話
側の音声信号を監視して通話状態を判定する通話状態判
定部と、送話側損失挿入手段の入力信号並びに受話側損
失挿入手段の出力信号から近端側の帰還利得を推定する
近端側帰還利得推定部と、受話側損失挿入手段の入力信
号並びに送話側損失挿入手段の出力信号から遠端側の帰
還利得を推定する遠端側帰還利得推定部と、近端側帰還
利得推定部及び遠端側帰還利得推定部の各推定値に基づ
いて閉ループに挿入すべき総損失量を算出する総損失量
算出部と、通話状態判定部の判定結果と総損失量算出部
の算出値に応じて送話側損失挿入手段及び受話側挿入損
失手段の各挿入損失量を決定する挿入損失量分配処理部
とを具備し、近端側帰還利得推定部は、送話側損失挿入
手段の入力信号の時間平均パワーを推定する第１の平均
パワー推定部、受話側損失挿入手段の出力信号の時間平
均パワーを推定する第２の平均パワー推定部、受話側損
失挿入手段の出力点から送話側損失挿入手段の入力点の
間の近端側帰還経路にて想定される最大遅延時間におけ
る第２の平均パワー推定部の出力の最小値を求める第１
の最小値算出部、第１の平均パワー推定部の出力値を第
１の最小値算出部の出力値で除算する第１の除算器を有
して通話状態判定部により受話状態と判定されたときに
のみ各部の処理を更新して成り、遠端側帰還利得推定部
は、受話側損失挿入手段の入力信号の時間平均パワーを
推定する第３の平均パワー推定部、送話側損失挿入手段
の出力信号の時間平均パワーを推定する第４の平均パワ
ー推定部、送話側損失挿入手段の出力点から受話側損失
挿入手段の入力点の間の遠端側帰還経路にて想定される
最大遅延時間における第４の平均パワー推定部の出力の
最小値を求める第２の最小値算出部、第３の平均パワー
推定部の出力値を第２の最小値算出部の出力値で除算す
る第２の除算部を有して通話状態判定部により送話状態
と判定されたときにのみ各部の処理を更新して成ること
を特徴とする拡声通話装置。1. A microphone that outputs a collected sound as an audio signal on a transmitting side, a first amplifying unit that amplifies an audio signal from the microphone, a speaker that sounds according to an audio signal on a receiving side, Second amplification means for amplifying the audio signal output to the speaker, transmission-side loss insertion means for inserting a predetermined amount of loss in the signal path on the transmission side, and loss of a predetermined amount in the signal path on the reception side. Receiving side loss insertion means to be inserted;
In a loudspeaker apparatus comprising: a receiving-side loss insertion unit and an insertion-loss-amount adjusting unit that adjusts an insertion loss amount of the transmitting-side loss-inserting unit, the insertion-loss-amount adjusting unit converts an audio signal on the transmitting side and a voice signal on the receiving side into A call state determination unit that monitors and determines a call state; and a near-end feedback gain estimation unit that estimates a near-end feedback gain from an input signal of the transmission-side loss insertion unit and an output signal of the reception-side loss insertion unit. A far-end feedback gain estimating unit for estimating a far-end feedback gain from an input signal of the receiving-side loss inserting unit and an output signal of the transmitting-side loss inserting unit; a near-end feedback gain estimating unit and a far-end feedback A total loss amount calculator for calculating a total loss amount to be inserted into the closed loop based on each estimated value of the gain estimator; and a transmitting side according to a determination result of the call state determination unit and a calculated value of the total loss amount calculation unit. Insertion loss of loss insertion means and receiver insertion loss means A near-end feedback gain estimating unit for estimating a time average power of an input signal of the transmitting-side loss inserting unit; A second average power estimator for estimating the time average power of the output signal of the loss insertion means, assumed on a near-end feedback path between an output point of the reception-side loss insertion means and an input point of the transmission-side loss insertion means; To determine the minimum value of the output of the second average power estimator during the maximum delay time
And a first divider for dividing the output value of the first average power estimating unit by the output value of the first average power estimating unit. The far-end feedback gain estimating unit estimates the time-average power of the input signal of the receiving-side loss inserting unit, and the transmitting-side loss inserting unit. A fourth average power estimating unit for estimating the time average power of the output signal of the receiver, the maximum assumed on the far-end feedback path between the output point of the transmitting-side loss inserting means and the input point of the receiving-side loss inserting means. A second minimum value calculator for obtaining the minimum value of the output of the fourth average power estimator during the delay time, and a second minimum value calculator for dividing the output value of the third average power estimator by the output value of the second minimum value calculator. When the call state is determined by the call state determination unit to have a division unit of 2 Hands-free communication device, characterized by comprising updating the process of each section only.
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、送話側の信号経路に所定量
の損失を挿入する送話側損失挿入手段と、受話側の信号
経路に所定量の損失を挿入する受話側損失挿入手段と、
受話側損失挿入手段及び送話側損失挿入手段の挿入損失
量を調整する挿入損失量調整手段とを備えた拡声通話装
置において、挿入損失量調整手段は、送話側並びに受話
側の音声信号を監視して通話状態を判定する通話状態判
定部と、送話側損失挿入手段の入力信号並びに受話側損
失挿入手段の出力信号から近端側の帰還利得を推定する
近端側帰還利得推定部と、受話側損失挿入手段の入力信
号並びに送話側損失挿入手段の出力信号から遠端側の帰
還利得を推定する遠端側帰還利得推定部と、近端側帰還
利得推定部及び遠端側帰還利得推定部の各推定値に基づ
いて閉ループに挿入すべき総損失量を算出する総損失量
算出部と、通話状態判定部の判定結果と総損失量算出部
の算出値に応じて送話側損失挿入手段及び受話側挿入損
失手段の各挿入損失量を決定する挿入損失量分配処理部
とを具備し、近端側帰還利得推定部は、送話側損失挿入
手段の入力信号の時間平均パワーを推定する第１の平均
パワー推定部、受話側損失挿入手段の出力信号の時間平
均パワーを推定する第２の平均パワー推定部、第２の平
均パワー推定部の出力を閉ループの最小伝達遅延時間以
上の時間だけ遅延させる第１の遅延部、第１の平均パワ
ー推定部の出力値を第１の遅延部の出力値で除算する第
１の除算器を有して通話状態判定部により受話状態と判
定されたときにのみ各部の処理を更新して成り、遠端側
帰還利得推定部は、受話側損失挿入手段の入力信号の時
間平均パワーを推定する第３の平均パワー推定部、送話
側損失挿入手段の出力信号の時間平均パワーを推定する
第４の平均パワー推定部、第４の平均パワー推定部の出
力を閉ループの最小伝達遅延時間以上の時間だけ遅延さ
せる第２の遅延部、第３の平均パワー推定部の出力値を
第２の遅延部の出力値で除算する第２の除算部を有して
通話状態判定部により送話状態と判定されたときにのみ
各部の処理を更新して成ることを特徴とする拡声通話装
置。2. A microphone that outputs a collected sound as an audio signal on the transmitting side, a first amplifying unit that amplifies an audio signal from the microphone, and a speaker that sounds in response to the audio signal on the receiving side. Second amplification means for amplifying the audio signal output to the speaker, transmission-side loss insertion means for inserting a predetermined amount of loss in the signal path on the transmission side, and loss of a predetermined amount in the signal path on the reception side. Receiving side loss insertion means to be inserted;
In a loudspeaker apparatus comprising: a receiving-side loss insertion unit and an insertion-loss-amount adjusting unit that adjusts an insertion loss amount of the transmitting-side loss-inserting unit, the insertion-loss-amount adjusting unit converts a voice signal of the transmitting side and a voice signal of the receiving side into a sound. A call state determination unit that monitors and determines a call state; and a near-end feedback gain estimation unit that estimates a near-end feedback gain from an input signal of the transmission-side loss insertion unit and an output signal of the reception-side loss insertion unit. A far-end feedback gain estimating unit for estimating a far-end feedback gain from an input signal of the receiving-side loss inserting unit and an output signal of the transmitting-side loss inserting unit; a near-end feedback gain estimating unit and a far-end feedback A total loss amount calculator for calculating a total loss amount to be inserted into the closed loop based on each estimated value of the gain estimator; and a transmitting side according to a determination result of the call state determination unit and a calculated value of the total loss amount calculation unit. Insertion loss of loss insertion means and receiver insertion loss means A near-end feedback gain estimating unit for estimating a time average power of an input signal of the transmitting-side loss inserting unit; A second average power estimator for estimating the time average power of the output signal of the loss insertion means, a first delay unit for delaying the output of the second average power estimator by a time equal to or longer than the minimum transmission delay time of the closed loop; A first divider for dividing the output value of the average power estimating unit by the output value of the first delay unit, and updating the processing of each unit only when it is determined that the telephone is in the receiving state by the call state determining unit. A far-end feedback gain estimator for estimating the time-average power of the input signal of the receiving-side loss insertion means, and a time-average power of the output signal of the transmitting-side loss insertion means. The fourth average power estimating unit, A second delay unit that delays the output of the power estimating unit by a time equal to or longer than the minimum transmission delay time of the closed loop, and a second division that divides the output value of the third average power estimating unit by the output value of the second delay unit A loudspeaker apparatus comprising: a unit for updating a process of each unit only when the communication state determination unit determines that the communication state is a transmission state.
ら人間の音声の音韻継続時間以上の所定時間内における
最小値を算出する第３及び第４の最小値算出部を近端側
帰還利得推定部並びに遠端側帰還利得推定部にそれぞれ
設け、第３及び第４の最小値算出部の出力値を近端側帰
還利得推定部並びに遠端側帰還利得推定部の推定値とし
て成ることを特徴とする請求項１記載の拡声通話装置。3. A third and fourth minimum value calculation unit for calculating a minimum value within a predetermined time equal to or longer than a phoneme duration of a human voice from output values of the first and second dividers at a near end. Provided in the side feedback gain estimator and the far end feedback gain estimator, and using the output values of the third and fourth minimum value calculators as the estimated values of the near end feedback gain estimator and the far end feedback gain estimator. The loudspeaker apparatus according to claim 1, wherein the loudspeaker apparatus comprises:
遠端側帰還経路の少なくとも何れか一方に設けられるエ
コーキャンセラと、適応フィルタの係数の収束値からエ
コーキャンセラを設けた側の帰還経路の利得を近似的に
算出する帰還利得近似算出部と、この帰還利得近似算出
部の出力値とエコーキャンセラを設けた側の帰還利得推
定部の帰還利得推定値との比を求める第３の除算器と、
第３の除算器の出力値に応じてエコーキャンセラを設け
た側の帰還利得推定部の帰還利得推定値の最新の値と以
前の値の何れか一方を選択して総損失量算出部に出力す
る帰還利得選択部とを備え、この帰還利得選択部は、第
３の除算器の出力値が所定の範囲内である場合には最新
の帰還利得推定値を選択し、第３の除算器の出力値が所
定の範囲を超えた場合には以前の帰還利得推定値を選択
して成ることを特徴とする請求項１記載の拡声通話装
置。4. An echo canceller provided with at least one of a near-end feedback path and a far-end feedback path having an adaptive filter, and a feedback path provided with an echo canceller based on a convergence value of coefficients of the adaptive filter. And a third division for calculating a ratio between the output value of the feedback gain approximation calculation unit and the feedback gain estimation value of the feedback gain estimation unit provided with the echo canceller. Vessels,
According to the output value of the third divider, one of the latest value and the previous value of the feedback gain estimation value of the feedback gain estimation unit provided with the echo canceller is selected and output to the total loss amount calculation unit. A feedback gain selection unit that selects the latest feedback gain estimation value when the output value of the third divider is within a predetermined range, 2. The loudspeaker apparatus according to claim 1, wherein when the output value exceeds a predetermined range, a previous feedback gain estimation value is selected.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP07292799A JP3580168B2 (en) | 1999-03-18 | 1999-03-18 | Loudspeaker |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| JP07292799A JP3580168B2 (en) | 1999-03-18 | 1999-03-18 | Loudspeaker |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| JP2000270089A true JP2000270089A (en) | 2000-09-29 |
| JP3580168B2 JP3580168B2 (en) | 2004-10-20 |
Family
ID=13503493
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| JP07292799A Expired - Fee Related JP3580168B2 (en) | 1999-03-18 | 1999-03-18 | Loudspeaker |
Country Status (1)
| Country | Link |
|---|---|
| JP (1) | JP3580168B2 (en) |
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP2254312A1 (en) | 2002-04-26 | 2010-11-24 | Nec Corporation | Portable telephone having a rotating display and two cameras |
| JP2013225747A (en) * | 2012-04-20 | 2013-10-31 | Panasonic Corp | Communication device |
| JP2014033372A (en) * | 2012-08-03 | 2014-02-20 | Panasonic Corp | Loudspeaker call device |
-
1999
- 1999-03-18 JP JP07292799A patent/JP3580168B2/en not_active Expired - Fee Related
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| EP2254312A1 (en) | 2002-04-26 | 2010-11-24 | Nec Corporation | Portable telephone having a rotating display and two cameras |
| JP2013225747A (en) * | 2012-04-20 | 2013-10-31 | Panasonic Corp | Communication device |
| JP2014033372A (en) * | 2012-08-03 | 2014-02-20 | Panasonic Corp | Loudspeaker call device |
Also Published As
| Publication number | Publication date |
|---|---|
| JP3580168B2 (en) | 2004-10-20 |
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