JPS5924582B2 - conference call method - Google Patents
conference call methodInfo
- Publication number
- JPS5924582B2 JPS5924582B2 JP1707577A JP1707577A JPS5924582B2 JP S5924582 B2 JPS5924582 B2 JP S5924582B2 JP 1707577 A JP1707577 A JP 1707577A JP 1707577 A JP1707577 A JP 1707577A JP S5924582 B2 JPS5924582 B2 JP S5924582B2
- Authority
- JP
- Japan
- Prior art keywords
- conference
- circuit
- stations
- loss
- terminal
- 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.)
- Expired
Links
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/56—Arrangements for connecting several subscribers to a common circuit, i.e. affording conference facilities
Landscapes
- Engineering & Computer Science (AREA)
- Multimedia (AREA)
- Signal Processing (AREA)
- Interconnected Communication Systems, Intercoms, And Interphones (AREA)
Description
【発明の詳細な説明】
本発明は、4線式電話回線、または私設電話回線を介し
て、センター装置を中心として、複数個の端末装置が接
続されるスピーカ・マイク式の会議電話方式に関するも
のである。DETAILED DESCRIPTION OF THE INVENTION The present invention relates to a speaker-microphone conference telephone system in which a plurality of terminal devices are connected to a central device via a four-wire telephone line or a private telephone line. It is.
従来、この種の会議電話方式は、第1図に示すように、
会議用ブリッジ回路1をセンター局におき、これを中心
にしてn回線の4線式電話回線を介してn個の会議電話
用端末装置2−1〜2−nが接続されて構成される。Conventionally, this type of conference telephone system, as shown in Figure 1,
A conference bridge circuit 1 is placed at a center station, and n conference telephone terminal devices 2-1 to 2-n are connected to the conference bridge circuit 1 via n four-wire telephone lines.
この端末装置2は、第2図に示すように、マイクロホン
4、マイクアンプ3a、送話可変損失回路3bおよびラ
ンプアンプ3cを経て、電話回線の送信側Tに接続され
ている。一方電話回線の受信側Rは、受信可変損失回路
3d、スピーカアンプ3eを介してスピーカ5に接続さ
れる。マイクアンプ3aの出力及び受話回線Rは音声ス
イッチ制御回路3fにも接続されている。いま、マイク
ロホン4に入力が印加されると、そのマイク出力信号は
マイクアンプ3aで増幅されて、音声スイッチ制御回路
3fに与えられる。音声スイッチ回路3fは、その直流
制御電圧により送話可変損失回路3bの損失を最小にし
(送話路オン状態)、同時に受話可変損失回路3dには
、ある一定の損失(以下そう大損失Lvと称す)を与え
る(受話路オフ状態)。逆に、受話回線Rに入力信号が
与えられた場合、受話可変損失回路3dの損失は最小と
なり、送話可変損失回路3bの損失は最大となる。この
ようにして、送話路および受話路の何れか一方にそう大
損失Lvを与えることによつて、スピーカとマイクロホ
ンの音響的結合損失に起因するハウリングの発生を防ぐ
ことができる。As shown in FIG. 2, this terminal device 2 is connected to a transmitting side T of a telephone line via a microphone 4, a microphone amplifier 3a, a variable transmission loss circuit 3b, and a lamp amplifier 3c. On the other hand, the receiving side R of the telephone line is connected to a speaker 5 via a receiving variable loss circuit 3d and a speaker amplifier 3e. The output of the microphone amplifier 3a and the receiving line R are also connected to the audio switch control circuit 3f. Now, when an input is applied to the microphone 4, the microphone output signal is amplified by the microphone amplifier 3a and given to the audio switch control circuit 3f. The voice switch circuit 3f uses its DC control voltage to minimize the loss in the transmitting variable loss circuit 3b (sending line on state), and at the same time minimizes the loss in the receiving variable loss circuit 3d (hereinafter referred to as large loss Lv). (reception path off state). Conversely, when an input signal is applied to the receiving line R, the loss of the receiving variable loss circuit 3d becomes minimum, and the loss of the sending variable loss circuit 3b becomes maximum. In this way, by providing a large loss Lv to either the transmitting path or the receiving path, it is possible to prevent the occurrence of howling due to acoustic coupling loss between the speaker and the microphone.
端末装置の送話路あるいは受話路において、このそう大
損失Lvはその通話路がオンの時とオフの時のレベル差
に相当し、一般には送話路においても受話路においても
同一の値に設定される。ハウリング防止のためには、こ
のそう大損失Lvは大きければ大きいほど良いが、一方
通話品質上の制限から゜ あまり大きくすると、音声を
大きなレベル差でオン・オフする結果、語の頭が消えて
しまう話頭切断が生じたり、わりこみ通話が困難になつ
たりして、通話品質が劣化する。このために、一般には
、通話品質の点から見てそう大損失Lvを35dB、以
下とする必要がある。一方、会議に参加する局数nが多
くなればなるほど、各局のスピーカとマイクロホンの音
響結合(まわりこみ)が、相加される結果、ハウリング
が発生し易くなり、各端末のそう入損失Lvを大きくし
て、ハウリングの発生を防止する必要があり、上記のよ
うに必然的に通話品質は劣化する。In the sending path or receiving path of a terminal device, this large loss Lv corresponds to the level difference when the speaking path is on and off, and is generally the same value for both the sending path and the receiving path. Set. In order to prevent howling, the larger the loss Lv is, the better; however, due to limitations on call quality, if it is set too high, the beginning of the word will disappear as the sound is turned on and off with a large level difference. The quality of the call deteriorates as the beginning of the conversation is cut off and it becomes difficult to make a call. For this reason, it is generally necessary to keep the large loss Lv to 35 dB or less from the viewpoint of speech quality. On the other hand, as the number n of stations participating in a conference increases, the acoustic coupling between the speakers and microphones of each station increases, making it easier for howling to occur, which increases the input loss Lv of each terminal. Therefore, it is necessary to prevent howling from occurring, and as mentioned above, the quality of the call inevitably deteriorates.
この参加局数nのそう入損失に与える影響は、1010
g10(n−1)(n=2、3、・・・・・・・・・)
で表現される。例えば、n=30では、Lv=35dB
が要求されるとすると、n−5ではL=35−1010
g10(29−4)=21dBとなる。会議電話システ
ムを設計するに当つて、各端末の音声スイツチ回路のそ
う入損失Lは、最大参加局数Nmaxに合わせてハウリ
ングの起きない値に設定されねばならない。しかし、実
際の運用面から見ると、全局より少ない局の参加によつ
て会議を行うことが多い。しかし、そう入損失は最大局
数Nmaxに対応したLvの最大値に設定されているの
で、実際NrrlaX以下の局で会議が行なわれると、
より小さいそう入損失でより良好な通話ができるにもか
かわらず、大きなそう入損失の状態で会議を行なうこと
になり、悪い通話品質の状態で会議を行なわねばならな
いという欠点があつた。なお、参加局数に合わせて、各
端末において、手動でそう入損失Lを設定する方法も考
えられるが、これでは、設定に手間どるばかりでなく、
誤つた設定によつて会議に混乱を生ずるおそれがある。
本発明の目的は、上記の欠点を除去し、常に、実際の会
議参加局数に応じて、良好な通話品質で会議を行うこと
のできる拡声式の会議電話方式を提供するにある。The influence of this number of participating stations n on the input loss is 1010
g10(n-1) (n=2, 3,...)
It is expressed as For example, when n=30, Lv=35dB
is required, then L=35-1010 for n-5
g10(29-4)=21 dB. When designing a conference telephone system, the input loss L of the audio switch circuit of each terminal must be set to a value that does not cause howling in accordance with the maximum number of participating stations Nmax. However, from an actual operational point of view, meetings are often held with the participation of fewer stations than all stations. However, since the input loss is set to the maximum value of Lv corresponding to the maximum number of stations Nmax, if a conference is actually held with stations below NrrlaX,
Even though it is possible to make a better call with a smaller input loss, the conference has to be held with a large input loss, and the conference has to be held with poor call quality. It is also possible to manually set the input loss L on each terminal according to the number of participating stations, but this would not only be time-consuming, but also
Incorrect settings may cause confusion in the meeting.
SUMMARY OF THE INVENTION An object of the present invention is to provide a loudspeaker conference telephone system that eliminates the above-mentioned drawbacks and allows conferences to be held with good call quality at all times, depending on the actual number of conference participants.
本発明によれば、音声スイツチ回路を用いたスピーカ・
マイク式の端末装置を複数の局ごとに設け、前記複数の
端末装置のそれぞれを4線式電話回線、または私設電話
回線を介してセンター局を中心に接続して、前記複数の
局との間に拡声式の会議通話を行う会議電話方式におい
て、前記複数の局のうちの会議参加局から送出されるそ
れぞれの制御信号を前記センター局の装置によつて受信
し、前記制御信号の数をカウントして前記会議参加局の
数に対応した制御信号をつくり、該制御信号を前記複数
の端末装置に送出することによつて、前記複数の端末装
置が前記参加局の数に応じた最適の音声スイツチ回路の
そう入損失を設定することを特徴とした会議電話方式が
得られる。According to the present invention, a speaker using an audio switch circuit
A microphone-type terminal device is provided for each of the plurality of stations, and each of the plurality of terminal devices is connected to the center station via a 4-wire telephone line or a private telephone line, and the communication between the plurality of stations is In a conference telephone system in which a conference call is made using loudspeaker, each control signal transmitted from a conference participating station among the plurality of stations is received by the device of the center station, and the number of the control signals is counted. By creating a control signal corresponding to the number of stations participating in the conference, and sending the control signal to the plurality of terminal devices, the plurality of terminal devices can generate optimal audio according to the number of stations participating in the conference. A conference telephone system characterized by setting the input loss of the switch circuit is obtained.
次に、図面を参照して、本発明の1実施例を詳細に説明
する。Next, one embodiment of the present invention will be described in detail with reference to the drawings.
第3図は、本発明による会議電話方式の端末装置をプロ
ツク図によつて小したもので、マイクロホン11の出力
はマイクロアンプ12をとおつて音声制御回路22と、
この音声制御回路22の直流出力電圧によつてオン、オ
フする送話音声ゲート回路13に加えられる。FIG. 3 shows a simplified block diagram of the conference telephone type terminal device according to the present invention.
The signal is applied to the transmitting voice gate circuit 13 which is turned on and off depending on the DC output voltage of the voice control circuit 22.
送話音声ゲート回路13の出力側はラインアンプ15を
介して4線式電話回線の送話側端子16に接続されてい
る。一方、4線式電話回線の受話側端子17の入力側は
、前記音声制御回路22によつてオン、オフされる受話
音声ゲート回路18に接続され、この出力側はスピーカ
アンプ20を介してスピーカ21に接続されている。ス
イツチ23は端末装置で操作できるように設けられたス
イツチ、あるいは接点を示し、多周波発振器24を駆動
するごとく接続される。多周波発振器24の出力側はバ
ツフアーアンプ25を介して送話側端子16に接続され
ている。受話側端子17は多周波受信器26にも接続さ
れており、その出力は損失制御回路27に加えられる。
前記送話音声ゲート回路13の入出力端および前記受話
音声ゲート回路18の入出力端には、それぞれ送話補助
可変損失回路14および受話補助可変損失回路19が接
続されており、これ等はともに損失制御回路27の直流
出力電圧によつて制御をうける。また、第4図は、本発
明による会議電話方式のセンター局に配置されるセンタ
ー装置をプロツク図によつて示したもので、4線式の電
話回線應1の受話側は端子1Rに、送話側は端子1Tに
、また屈2回線の受話側は端子2R、送話側は端子2T
に接続される。The output side of the outgoing voice gate circuit 13 is connected via a line amplifier 15 to a outgoing end terminal 16 of a four-wire telephone line. On the other hand, the input side of the receiving side terminal 17 of the 4-wire telephone line is connected to the receiving audio gate circuit 18 which is turned on and off by the audio control circuit 22, and the output side of this terminal is connected to the receiving audio gate circuit 18 via the speaker amplifier 20. 21. The switch 23 represents a switch or contact provided to be operated by the terminal device, and is connected to drive the multi-frequency oscillator 24. The output side of the multi-frequency oscillator 24 is connected to the transmitting side terminal 16 via a buffer amplifier 25. The receiver terminal 17 is also connected to a multifrequency receiver 26 , the output of which is applied to a loss control circuit 27 .
A transmitting auxiliary variable loss circuit 14 and a receiving auxiliary variable loss circuit 19 are connected to the input and output terminals of the transmitting audio gate circuit 13 and the receiving audio gate circuit 18, respectively. It is controlled by the DC output voltage of the loss control circuit 27. FIG. 4 is a block diagram showing the center equipment installed in the center office of the conference telephone system according to the present invention. The talking side is connected to terminal 1T, the receiving side of the 2nd line is connected to terminal 2R, and the sending side is connected to terminal 2T.
connected to.
以下同様に、黒n回路の受話側は端子NRに、送話側は
端子NTに接続され、これらn個の4線式電話回線のそ
れぞれ一方は会議用ブリツジ回路51に収容されている
。多周波受信器52−1〜52−nはn組の4線式電話
回線のそれぞれ受話側に接続されており、これらの出力
はオア回路53に与えられる。Similarly, the receiving side of the black n circuit is connected to the terminal NR, and the transmitting side is connected to the terminal NT, and one of each of these n four-wire telephone lines is accommodated in the conference bridge circuit 51. The multi-frequency receivers 52-1 to 52-n are connected to the receiving end of n sets of four-wire telephone lines, and their outputs are given to an OR circuit 53.
オア回路53の出力61はn進カウンタ54に加えられ
、nビツトよりなる並列出力62−1〜62一nを出力
する。これ等の出力は多周波発振器制御回路55に接続
され、この出力リード線63−1〜63−16によつて
抽出された16種類の制御信号が多周波発振器56に供
給される。多周波発振器56の出力64は、バツフアー
アンプ591〜59−nを介してそれぞれn組の4線式
電話回線の送話側に接続される。一方、オア回路53の
出力61はタイマー回路57に与えられ、その出力は多
周波発振器56を駆動すべく、その入力に加えられる。
なお、スイツチ58はセンター装置のキーで、n進カウ
ンタ54のりセツト入力に接続されている。次に、本発
明による会議電話方式を構成する上記端末装置とセンタ
ー装置の綜合的な動作について説明する。The output 61 of the OR circuit 53 is added to the n-ary counter 54, which outputs parallel outputs 62-1 to 62-n each consisting of n bits. These outputs are connected to a multi-frequency oscillator control circuit 55, and 16 types of control signals extracted through the output lead wires 63-1 to 63-16 are supplied to the multi-frequency oscillator 56. The output 64 of the multi-frequency oscillator 56 is connected to the transmitting side of n sets of four-wire telephone lines via buffer amplifiers 591 to 59-n, respectively. On the other hand, the output 61 of the OR circuit 53 is given to a timer circuit 57, and its output is added to the input of the multi-frequency oscillator 56 in order to drive it.
The switch 58 is a key of the center device and is connected to the input of the n-ary counter 54. Next, the overall operation of the terminal device and center device that constitute the conference telephone system according to the present invention will be explained.
先づ、第3図において、スイツチ23は、端末装置の使
用者が電源を入れると連動して動作するか、あるいは、
別に設けられた起動キーを押すと動作する。スイツチ2
3が閉じると、多周波発振器24が起動し、ある特定の
各端末に共通な多周波信号が、一定時間、例えば100
mS、バツフアーアンプ25を介して電話回線の送話側
端子16へ送出される。この端未起動信号は、システム
の参加局のそれぞれの端末から、ある時間間隔をもつた
同一の多周波信号が4線式電話回線を介してそれぞれセ
ンター装置の端子1R,2R,・・・・・・・・・,N
Rに与えられる。これらの多周波信号は各電話回線に対
応する多周波受信器52−1〜52−nで受信、検出さ
れ、これらの出力60−1〜60−nに単一のパルス信
号を導いて、それぞれオア回路53に加えられる。各端
末からの制御信号は、各端末が同時に起動信号を出す確
率は非常に低いので、一般に、ある時間間隔をもつて受
信される。そのために、オア回路53の出力には、参加
局数をiとすれば、それと同じ数のi個のパルス列が表
われ、n進計数回路54に入力される。また、オア回路
53の出力61は、ある時間幅のパルス(例えば100
msec)を発生するところのタイマー回路57にも印
加されているので、タイマー回路57の入力にパルスが
与えられるごとに、出力に100msecの単一パルス
が発生し、それによつて多周波発信器56を駆動する。
カウンター回路54のn本の並夕1拙力62一1〜62
−nには会議参加局数1に応じたパルスが表われる。こ
れらの出力が多周波発振器制御回路55に与えられると
、その16本の出力63一1〜63−16のうち、1本
がカウンター54の計数値、即ち、参加局数1に対応し
て、論理“O゛レベルとなり、多周波発振回路56の中
の16組の2周波発振回路の1つを選択する。前述のよ
うに、多周波発振回路56は、各端末からの制御パルス
が受信されるごとに、100msecのパルスを発生す
るタイマー回路57によつて駆動され、その出力64に
は、参加局数1に対応した多周波信号(100msec
)が抽出される。この信号は、n個のバツフアーアンプ
59−1〜59−nを介して、全てのn個の端末へ、4
線式電話回線の送話側端子1T,2T,・・・・・・・
・・,NTを介して一斉に送出される。多周波発振回路
56は16種類の2周波組合せ信号を発生できるので、
各々の2周波組合せ信号に前述の端末装置のそう入損失
Lを1対1に対応させれば、参加局数に応じて、16種
類のそう入損失Lvを制御できることになる。First, in FIG. 3, the switch 23 operates in conjunction with when the user of the terminal device turns on the power, or
It works by pressing a separate activation key. switch 2
3 closes, the multi-frequency oscillator 24 is started, and the multi-frequency signal common to each specific terminal is emitted for a certain period of time, for example, 100
mS, and is sent to the transmitting side terminal 16 of the telephone line via the buffer amplifier 25. This end inactivation signal is the same multi-frequency signal with a certain time interval from each terminal of the participating stations of the system via the 4-wire telephone line to the terminals 1R, 2R, . . . of the center equipment, respectively. ...,N
given to R. These multi-frequency signals are received and detected by multi-frequency receivers 52-1 to 52-n corresponding to each telephone line, and a single pulse signal is guided to these outputs 60-1 to 60-n, respectively. It is added to the OR circuit 53. Control signals from each terminal are generally received at certain time intervals since the probability that each terminal issues an activation signal at the same time is very low. Therefore, if the number of participating stations is i, the same number i pulse trains appear at the output of the OR circuit 53 and are input to the n-ary counting circuit 54. Further, the output 61 of the OR circuit 53 is a pulse of a certain time width (for example, 100
msec), so that every time a pulse is applied to the input of the timer circuit 57, a single pulse of 100 msec is generated at the output, thereby causing the multifrequency oscillator 56 to generate a single pulse of 100 msec. to drive.
Counter circuit 54 of n lines 62-1 to 62
A pulse corresponding to the number of stations participating in the conference (1) appears at -n. When these outputs are given to the multi-frequency oscillator control circuit 55, one of the 16 outputs 63-1 to 63-16 corresponds to the count value of the counter 54, that is, the number of participating stations is 1, The logic becomes "O" level, and one of the 16 sets of two-frequency oscillation circuits in the multi-frequency oscillation circuit 56 is selected.As described above, the multi-frequency oscillation circuit 56 receives control pulses from each terminal. It is driven by a timer circuit 57 that generates a 100 msec pulse every time the station is connected, and its output 64 contains a multi-frequency signal (100 msec pulse) corresponding to the number of participating stations (1).
) is extracted. This signal is transmitted to all n terminals via n buffer amplifiers 59-1 to 59-n.
Sending side terminals of wire telephone line 1T, 2T,...
..., are sent out all at once via NT. Since the multi-frequency oscillation circuit 56 can generate 16 types of two-frequency combination signals,
If each two-frequency combination signal is associated with the input loss L of the terminal device described above on a one-to-one basis, 16 types of input loss Lv can be controlled depending on the number of participating stations.
実際的には、Lv−6、8、10、・・・・・・・・・
、36dBの段階で設定することができれば十分である
。また、会議が終了するとセンター装置のりセツトキー
58を押すことにより、カウンター回路54はりセツト
される。一方、上述のごとく、各端末へー斉に送出され
るそう入損失制御信号(2周波信号)は、4線式電話回
線を介して、第3図に示す端末装置の端子17に与えら
れる。In reality, Lv-6, 8, 10, etc.
, it is sufficient to be able to set it in steps of 36 dB. Further, when the conference ends, the counter circuit 54 is reset by pressing the center device reset key 58. On the other hand, as described above, the input loss control signal (two-frequency signal) sent simultaneously to each terminal is applied to the terminal 17 of the terminal device shown in FIG. 3 via a four-wire telephone line.
この制御信号は多周波受信器26で受信検出される。多
周波受信器26の出力28には、受信2周波信号に対応
したコードが得られる。この出力コードをうけた損失制
御回路27は、そのコードに対応する直流電圧を発生す
る。この制御出力29は、FETダイオード等よりなる
送話補助可変損失回路14と受話補助可変損失回路19
に与えられ、制御直流電圧に応じてこれらの損失Lを設
定すべく制御する。送話音声ゲート回路13と受話音声
ゲート回路18は、FETlダイオード等により構成さ
れる音声制御回路22の出力直流電圧によつてオン、オ
フされ、送話(または受話)がオンのとき、受話(また
は二エ!!叫:=:=J回路19は、それぞれ送話音声
ゲート回路13および受話音声ゲート回路18に並列に
接続されているので、これらの損失が前述のそう入損授
Lを与えることになる。This control signal is received and detected by the multi-frequency receiver 26. At the output 28 of the multi-frequency receiver 26, a code corresponding to the received two-frequency signal is obtained. The loss control circuit 27 receiving this output code generates a DC voltage corresponding to the code. This control output 29 is transmitted through a transmitting auxiliary variable loss circuit 14 and a receiving auxiliary variable loss circuit 19 consisting of FET diodes and the like.
The loss L is controlled to be set according to the control DC voltage. The transmitting voice gate circuit 13 and the receiving voice gate circuit 18 are turned on and off by the output DC voltage of the voice control circuit 22 composed of FETl diodes, etc., and when the transmitting (or receiving) is on, the receiving (or Or 2e!! Shout:=:=J circuit 19 is connected in parallel to transmitting voice gate circuit 13 and receiving voice gate circuit 18, respectively, so these losses give the above-mentioned input loss loss L. It turns out.
本発明によれば、上述のように、設置された局数のうち
会議参加局数に対応して、各端末の音声スイツチのそう
入損失をセンターからの制御信号により自動的に設定で
きるので、常に最良の状態で、良好な拡声式の会議通話
が可能となる。According to the present invention, as described above, the input loss of the audio switch of each terminal can be automatically set by the control signal from the center according to the number of stations participating in the conference out of the number of installed stations. This makes it possible to always have conference calls with good amplification under the best conditions.
一例をあげて説明すると、会議電話システムの局数Nm
ax−30とすると、通常の会議室を用いた場合、本シ
ステムがハウリングを生じないためには、各端末のそう
入損失Lvは、23dB以上に設定する必要がある。L
−25dBにセツトするとすれば、話頭切断等が生じ、
通話性能は満足ではない。ところが、このように30局
まで接続可能なシステムにおいて、会議参加局数が5局
の場合を考えると、ハウリングを防ぐためには、各端末
の音声スイツチ回路のそう人損失Lvは14dBにセツ
トすれば良く、L−14dBであれば、通話路のオン、
オフのレベル差が少ないので、話頭切断もほとんど感じ
られず、高品質の通話が可能となる。本発明によるシス
テムにおいて、局数30の場合Lv−25dBにセツト
されていても、実際の参加局数が5の場合には、センタ
ーからの制御信号により、自動的にLv−14dBにセ
ツトされるので、良好な音声品質の会議通話を行なうこ
とができる。もしも、本発明のように、端末のそう入損
失Lvが参加局数に応じて変化しないとすれば、上述の
例では、参加局数が5局となつたにもかかわらず最大局
数30に対応したLv一25dBで通話することになり
、高品質の会議通話が期待できない。本発明は以上説明
したように、会議参加局数に対応して各端末の音声スイ
ツチ回路のそう入損失を、センターからの制御信号によ
つて自動的に設定することができるから、会議参加局数
に応じた最良の通話品質で会議通話を行うことができる
点において、その得られる効果は大きい。To explain with an example, the number of stations in a conference telephone system is Nm.
Assuming ax-30, when using a normal conference room, the input loss Lv of each terminal needs to be set to 23 dB or more in order for this system to not cause howling. L
If you set it to -25dB, the beginning of the conversation will be cut off, etc.
Call performance is not satisfactory. However, in a system where up to 30 stations can be connected, if the number of stations participating in the conference is 5, in order to prevent howling, the noise loss Lv of the audio switch circuit of each terminal should be set to 14 dB. Good, if it is L-14dB, the communication path is on.
Since there is little difference in the off level, there is almost no noticeable disconnection at the beginning of the conversation, and high-quality calls are possible. In the system according to the present invention, even if Lv is set to -25 dB when the number of stations is 30, when the actual number of participating stations is 5, it is automatically set to Lv -14 dB by a control signal from the center. Therefore, conference calls with good voice quality can be made. If the input loss Lv of a terminal does not change according to the number of participating stations as in the present invention, in the above example, the maximum number of stations would be 30 even though the number of participating stations was 5. You will have to make a call at 25dB below the supported Lv, so you cannot expect a high-quality conference call. As explained above, the present invention allows the input loss of the audio switch circuit of each terminal to be automatically set according to the number of conference participating stations using the control signal from the center. The effect obtained is significant in that conference calls can be made with the best call quality depending on the number of conference calls.
第1図は従来の会議電話方式の1例を示す構成図、第2
図は第1図の方式における端末装置の1例を示すプロツ
ク図、第3図は、本発明による一実施例を説明するため
の、会議用端末装置のプロツク図、第4図は本発明によ
る一実施例を説明するための、会議電話用センター装置
のプロック図である。
図において、11はマイクロホン、13は送話音声ゲー
ト回路、14は送話補助可変損失回路、18は受話音声
ゲート回路、19は受話補助可変損失回路、21はスピ
ーカ、22は音声制御回路、24は多周波発振器、26
は多周波受信器、27は損失制御回路、51は会議用ブ
リッジ回路、52−1〜52−nは多周波受信器、53
はオア回路、54はn進カウンタ、55は多周波発振器
制御回路、56は多周波発振回路である。Figure 1 is a configuration diagram showing an example of a conventional conference telephone system;
1 is a block diagram showing an example of a terminal device according to the method of FIG. 1, FIG. 3 is a block diagram of a conference terminal device for explaining an embodiment according to the present invention, and FIG. FIG. 2 is a block diagram of a conference call center device for explaining one embodiment. In the figure, 11 is a microphone, 13 is a transmitting audio gate circuit, 14 is a transmitting auxiliary variable loss circuit, 18 is a receiving audio gate circuit, 19 is a receiving auxiliary variable loss circuit, 21 is a speaker, 22 is an audio control circuit, 24 is a multi-frequency oscillator, 26
is a multi-frequency receiver, 27 is a loss control circuit, 51 is a conference bridge circuit, 52-1 to 52-n are multi-frequency receivers, 53
is an OR circuit, 54 is an n-ary counter, 55 is a multi-frequency oscillator control circuit, and 56 is a multi-frequency oscillation circuit.
Claims (1)
末装置を複数の局ごとに設け、前記複数の端末装置のそ
れぞれを4線式電話回線、または私設電話回線を介して
センター局を中心に接続して、前記複数の局との間に拡
声式の会議通話を行う会議電話方式において、前記複数
の局のうちの会議参加局から送出されるそれぞれの制御
信号を前記センター局の装置によつて受信し、前記制御
信号の数をカウントして、前記会議参加局の数に対応し
た制御信号をつくり、該制御信号を前記複数の端末装置
に送出することによつて、前記複数の端末装置が前記会
議参加局の数に応じた最適の音声スイッチ回路のそう入
損失を設定することを特徴とした会議電話方式。1 A speaker-microphone type terminal device using an audio switch circuit is installed at each of a plurality of stations, and each of the plurality of terminal devices is connected to the center station via a four-wire telephone line or a private telephone line. In a conference telephone system in which a loudspeaker conference call is made between the plurality of stations, each control signal sent from a conference participating station among the plurality of stations is received by the device of the center station. Then, by counting the number of control signals, creating a control signal corresponding to the number of stations participating in the conference, and transmitting the control signal to the plurality of terminal devices, the plurality of terminal devices A conference telephone system characterized by setting the optimum input loss of a voice switch circuit according to the number of stations participating in the conference.
Priority Applications (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
JP1707577A JPS5924582B2 (en) | 1977-02-21 | 1977-02-21 | conference call method |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
JP1707577A JPS5924582B2 (en) | 1977-02-21 | 1977-02-21 | conference call method |
Publications (2)
Publication Number | Publication Date |
---|---|
JPS53102604A JPS53102604A (en) | 1978-09-07 |
JPS5924582B2 true JPS5924582B2 (en) | 1984-06-11 |
Family
ID=11933850
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
JP1707577A Expired JPS5924582B2 (en) | 1977-02-21 | 1977-02-21 | conference call method |
Country Status (1)
Country | Link |
---|---|
JP (1) | JPS5924582B2 (en) |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JPH02141791U (en) * | 1989-04-28 | 1990-11-29 |
-
1977
- 1977-02-21 JP JP1707577A patent/JPS5924582B2/en not_active Expired
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JPH02141791U (en) * | 1989-04-28 | 1990-11-29 |
Also Published As
Publication number | Publication date |
---|---|
JPS53102604A (en) | 1978-09-07 |
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