JPS62154839A - Conference telephone equipment - Google Patents

Conference telephone equipment

Info

Publication number
JPS62154839A
JPS62154839A JP29439485A JP29439485A JPS62154839A JP S62154839 A JPS62154839 A JP S62154839A JP 29439485 A JP29439485 A JP 29439485A JP 29439485 A JP29439485 A JP 29439485A JP S62154839 A JPS62154839 A JP S62154839A
Authority
JP
Japan
Prior art keywords
loss
flop
circuit
flip
control signal
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.)
Pending
Application number
JP29439485A
Other languages
Japanese (ja)
Inventor
Toshiyuki Tachibana
橘 敏幸
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Toa Corp
Toa Tokushu Denki KK
Original Assignee
Toa Electric Co Ltd
Toa Tokushu Denki KK
Priority date (The priority date 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 date listed.)
Filing date
Publication date
Application filed by Toa Electric Co Ltd, Toa Tokushu Denki KK filed Critical Toa Electric Co Ltd
Priority to JP29439485A priority Critical patent/JPS62154839A/en
Publication of JPS62154839A publication Critical patent/JPS62154839A/en
Pending legal-status Critical Current

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  • Interconnected Communication Systems, Intercoms, And Interphones (AREA)

Abstract

PURPOSE:To improve the SN ratio and articulation of a transmitting signal by reducing only the loss quantity of a variable loss circuit and suppressing the reduction of the loss quantity of variable loss circuits in other terminal equipments when a detecting output signal is supplied to a flip flop (FF) as a reset signal. CONSTITUTION:At first, voice is initially reached and inputted to a transmitter 5 connected to a terminal equipment 4 arranged on the nearest position to the initial speaker, an output signal from a sound switch circuit 19 is turned to the 'L' level and a pre-FF 23 and a post-FF 23 are simultaneously inverted to the reset state. Consequently, the loss quantity of the variable loss circuit 16 is controlled from 10dB e.g. to 0dB, a control signal line 12 is turned to the 'L' level and all pre-FFs 22 and post-FFs 23 in other terminal equipments are turned to the reset state.

Description

【発明の詳細な説明】 (産業上の利用分野) この発明は、電話回線を通して互いに離れた場所にいる
複数人が、電話会議をするために使われる会議通話装置
に関する。
DETAILED DESCRIPTION OF THE INVENTION (Field of Industrial Application) The present invention relates to a conference call device used for a telephone conference between a plurality of people who are located apart from each other through a telephone line.

ぐ従来の技術) 従来送話用マイクロホンが接続された端末81を一方の
会議室等に複数台設置し、各マイクロホンで収音した送
話信号を加算して電話回線に送出することにより、他方
の会議室等に設置された端末装置との間で相互に通話す
るようにした会議通話装置があった。
Conventional technology) By installing a plurality of terminals 81 to which conventional transmitting microphones are connected in one conference room, etc., and adding the transmitting signals picked up by each microphone and transmitting the sum to the telephone line, There was a conference call device that was designed to communicate with terminal devices installed in a conference room or the like.

(発明が解決しようとする問題点) ところが、複数個のそれぞれのマイクロホンで収音した
騒音レベルが加算累積されることにより、送話信号のS
 / N比が悪(なっていた。また、送話者である音源
から距離的に離れて同一室内に設置された複数個のマイ
クロホンに到達する音波の経路がそれぞれ異なり、直接
音だけでなく反射音も収音しているため、これら位相差
および時間差のある音声信号を単純加算すると、反響や
残響の大きい送話信号となって明瞭度が下がり、受話側
で正確な送話内容が聞き取りにくくなるという問題点が
あった。
(Problem to be Solved by the Invention) However, as the noise level collected by each of the plurality of microphones is added and accumulated, the S of the transmission signal is
/ The N ratio was poor.Also, the paths of sound waves reaching multiple microphones installed in the same room at a distance from the sound source (speaker) are different, and there is not only direct sound but also reflected sound. Since sound is also collected, simply adding up these audio signals with phase and time differences results in a transmitted signal with large echoes and reverberations, reducing clarity and making it difficult for the receiver to hear the exact content of the transmitted message. There was a problem with that.

(問題点を解決するための手段) この発明は、上記従来の問題点を解消するために成され
たもので、その目的は、複数個のマイクロホンのうち同
時に動作するマイクロホンの数を実質的に減らすように
することにより、送話信号のS/N比および明[IF1
!を向上させることの出来る会&?iTi話装置を提供
することにある。
(Means for Solving the Problems) The present invention has been made to solve the above conventional problems, and its purpose is to substantially reduce the number of microphones that operate simultaneously among a plurality of microphones. By reducing the S/N ratio of the transmitting signal and the brightness [IF1
! A meeting that can improve the &? The purpose of the present invention is to provide an iTi talking device.

以下図面に示す実施例に基づいて、詳細に説明する。第
1図は会議電話装置のシステム構成図であり、電話回線
1に接続された1装[2に、接続ケーブル3で複数台の
端末装置4が順次接続され、端末装置4のそれぞれには
2個一対の双指向性を有するマイクロホンからなる送話
器5が取付けられるとともに、イヤホンまたはスピーカ
からなる4個の受話器6が接続されている。
The present invention will be described in detail below based on the embodiments shown in the drawings. FIG. 1 is a system configuration diagram of a conference telephone device, in which a plurality of terminal devices 4 are sequentially connected to one device [2] connected to a telephone line 1 with a connecting cable 3, and each terminal device 4 has two A transmitter 5 consisting of a pair of bidirectional microphones is attached, and four receivers 6 consisting of earphones or speakers are connected.

第2図に示すように、主装置2は2線4線変換回路7、
これに接続された受話信号用増幅器8と送話信号用混合
増幅器9をそなえている。接続ケーブル3は増幅器8の
出力側に接続された受話信号線10、混合増幅器90入
力側に接続された送話信号線11、制御信号線12をそ
なえており、この接続ケーブル3に各々の端末装@4が
共通に並列接続されている。端末装置4は、受話器6を
接続端子13へ接続または分離することに連動して、受
話器6と同じ側に配置された送話器5の動作をオンまた
はオフ制御するスイッチ14と、送話器5用の差動増幅
器15と、FETなどで構成され送話信号の損失Mを可
変するための可変損失回路16と、送話信号を送話信号
線11に送出するための出力増幅器17と、混合用抵抗
器18とが直列に接続されている。そして、差動増幅器
15の出力イd号を整流平滑してスレッシュホールド・
レベルと比較しこれを上回ったときアクティブ・レベル
の信号を出力する音声スイッチ回路19と、この出力信
号および制御信号線12から供給される制御信号の状態
に応じて、可変損失回路16の損失量および制御信号線
の制御信号を制御するための′#11御回路20とをそ
なえている。なお、音量調節!a21で受話器6の受話
音量が個別に調節されるようになっている。
As shown in FIG. 2, the main device 2 includes a 2-wire 4-wire conversion circuit 7,
A receiving signal amplifier 8 and a transmitting signal mixing amplifier 9 are connected to this. The connection cable 3 has a reception signal line 10 connected to the output side of the amplifier 8, a transmission signal line 11 connected to the input side of the mixing amplifier 90, and a control signal line 12. The devices @4 are commonly connected in parallel. The terminal device 4 includes a switch 14 that controls on/off operation of a transmitter 5 disposed on the same side as the receiver 6 in conjunction with connecting or disconnecting the receiver 6 to the connection terminal 13; 5 differential amplifier 15, a variable loss circuit 16 composed of FETs, etc., for varying the loss M of the transmission signal, and an output amplifier 17 for sending the transmission signal to the transmission signal line 11, A mixing resistor 18 is connected in series. Then, the output id of the differential amplifier 15 is rectified and smoothed to create a threshold value.
The audio switch circuit 19 outputs an active level signal when the level exceeds the level, and the loss amount of the variable loss circuit 16 is determined according to the state of this output signal and the control signal supplied from the control signal line 12. and a #11 control circuit 20 for controlling the control signal of the control signal line. In addition, you can adjust the volume! The listening volume of the handset 6 is individually adjusted at step a21.

制御回路20は、第3図に示すように、2段に接続され
たR3型フリップフロップ22・23により音源に最も
近い端末装置4の可変損失回路16の損失を少なくし、
他の端末装v!14の可変損失回路16の損失が小さく
ならない種制御する優先順位回路を構成しており、初段
フリップフロップ22のセット信号入力端子Sにプルア
ップ抵抗器24を介して音声スイッチ回路19の出力側
が接続され、リセット信号入力端子Rに制御信号線12
が接続されている。このi、II if回路20は正論
理で良いが、負論理すなわち信号が“L°′ (ロー)
レベルでアクティブ動作状態となる場合について例示し
ている。初段フリップフロップ22の出力端子Qと可は
、後段フリップフロップ23の入力端子RとSにそれぞ
れ接続され、後段フリップフロップの出力端子Qがイン
バータ用トランジスタ25、抵抗器26を介して制御信
号線12に接続されるとともに、可変損失回路16に損
失量制皿信号を供給するようになっている。後段フリッ
プフロップ23を省略して初段フリップフロップ22だ
けで構成するようにしても良い。
As shown in FIG. 3, the control circuit 20 reduces the loss of the variable loss circuit 16 of the terminal device 4 closest to the sound source by using R3 type flip-flops 22 and 23 connected in two stages.
Other terminal equipment v! The output side of the audio switch circuit 19 is connected to the set signal input terminal S of the first stage flip-flop 22 via a pull-up resistor 24. and the control signal line 12 is connected to the reset signal input terminal R.
is connected. This i, II if circuit 20 may be of positive logic, but it may be negative logic, that is, the signal is "L°'(low)".
An example is shown in which the active operating state is reached at the level. The output terminals Q and Q of the first stage flip-flop 22 are connected to the input terminals R and S of the second stage flip-flop 23, respectively, and the output terminal Q of the second stage flip-flop is connected to the control signal line 12 via an inverter transistor 25 and a resistor 26. The loss amount control signal is supplied to the variable loss circuit 16. The second stage flip-flop 23 may be omitted and the first stage flip-flop 22 may be used alone.

く作用) このように構成されており、端末装置4は、例えば会議
参加入数や会議室の大きさ等に合わせて同一室内の机上
に複数台設置した状態で使用される。第3図において、
いずれの端末装置20の送話器5へも音声入力が無い初
期状態では、初段フリップフロップ22はリセット状態
、すなわら端子百と石が’ l−1”レベルで端子Qが
゛′L°゛レベルとなり、後段フリップフロップ23が
リセット状態、すなわち端子互とQが゛L°゛レベルで
端子百とQが゛トビレベルとなり、可変損失回路16の
損失量が例えば10dBに、トランジスタ25のコレク
タ電極と制御信号線12と初段フリップフロップ22の
端子Rがともに1」゛レベルになっている。先ず、最初
の送話者すなわち音源に最も近い位置にある1台の端末
装置4に接続された送話器5に最初に音声が到達し入力
され、その音声スイッチ回路1つの出力信号が゛Lパレ
ベルになって、初段フリップフロップ22と、後段フリ
ップフロップ23が共にセット状態にそれぞれ反転する
。その結果、可変損失回路16の損失量が例えば10d
BからOdBに制御されるとともに、制御信号線12が
L”レベルになって、その他の端末装置4の初段フリッ
プフロップ22と後段フリップフロップ23の全てをリ
セット状態にする。従って、その他の端末装置4では送
話15で収音された音声信号や騒音信号が、可変損失回
路16で減衰さ才られ、送話信号線11に殆ど送出され
ない入力禁止状態になる。なお、このとき、最初にセッ
トされたフリップフロップ22の端子K El −’ 
L  レベルこはっか、m士ろ、にバ)w−Lレベルの
ときは、端子Q  、Qは共に“1−ド°レベルとなる
ので、後段フリップフロップ23はリセットされること
はない。このように、最初の送話開始者に最も近い端末
装置4に接続された送話器5からだけの信号が検出され
優先的に選択されて送話信号線11に送出され、その他
の端末装置4に接続された送話器5からの騒音信号は殆
ど加算累積されない。だから、同時にオン状態になる送
話器5の数が実質的に減少することに伴って、送話信号
のS/N比は向上するとともに、残響音や反W音が減少
することにより送話信号の明瞭度が向上する。
With this structure, a plurality of terminal devices 4 are installed on a desk in the same room depending on the number of participants in the conference, the size of the conference room, etc. In Figure 3,
In the initial state where there is no audio input to the transmitter 5 of any terminal device 20, the first stage flip-flop 22 is in a reset state, that is, the terminals 10 and 1 are at the 'l-1' level and the terminal Q is at the 'L' level. level, the subsequent flip-flop 23 is in a reset state, that is, the terminals 100 and Q are at the ``L°'' level, the terminals 100 and Q are at the ``tobi'' level, and the loss amount of the variable loss circuit 16 is, for example, 10 dB, and the collector electrode of the transistor 25 is in the reset state. The control signal line 12 and the terminal R of the first stage flip-flop 22 are both at the 1'' level. First, the voice first reaches and is input to the transmitter 5 connected to one terminal device 4 located at the position closest to the first speaker, that is, the sound source, and the output signal of one of the voice switch circuits is ゛L. At the same level, both the first-stage flip-flop 22 and the second-stage flip-flop 23 are inverted to the set state. As a result, the amount of loss of the variable loss circuit 16 is, for example, 10d.
B to OdB, the control signal line 12 goes to L'' level, and all of the first-stage flip-flops 22 and second-stage flip-flops 23 of the other terminal devices 4 are reset. In step 4, the voice signal and noise signal picked up by the transmitter 15 are attenuated by the variable loss circuit 16, and almost no input is sent to the transmitter signal line 11. The terminal K El −' of the flip-flop 22
When the L level is at the w-L level, both the terminals Q and Q are at the "1-de level", so the subsequent flip-flop 23 is not reset. Then, a signal only from the transmitter 5 connected to the terminal device 4 closest to the first person who initiated the transmission is detected, selected preferentially, and sent to the transmission signal line 11, and transmitted to the other terminal devices 4. The noise signals from the connected transmitters 5 are hardly added and accumulated.Therefore, as the number of transmitters 5 that are turned on at the same time is substantially reduced, the S/N ratio of the transmitted signal is reduced. At the same time, the clarity of the transmitted signal is improved by reducing reverberation and anti-W sound.

次に、送話中の送話器5への音声入力を止めると、その
端末装置4の音声スイッチ回路1つの出力信号がH”レ
ベルとなって、全てのフリップフロップ回路22・23
が元の初期状態に戻って、いづれの端末@置4からでも
割り込み送話可能な状態になる。
Next, when the audio input to the transmitter 5 during transmission is stopped, the output signal of one audio switch circuit of the terminal device 4 becomes H" level, and all the flip-flop circuits 22 and 23
returns to its original initial state and becomes ready for interrupt calls from any terminal @4.

(発明の効果) 上述のようにこの発明によれば、同一室内に設置された
複数個の送話器のうち、送話者に最も近い位置にある端
末装置の可変損失回路の損失を少なくし、他の端末装置
の可変損失回路の損失が大きい状態に自動制御すること
によって、オン状態にある送話器の数を実質的に減らす
ことができ、送話信号のS/N比および明瞭度を向上さ
せることができるという顕著な効果が得られる。
(Effects of the Invention) As described above, according to the present invention, it is possible to reduce the loss in the variable loss circuit of the terminal device located closest to the transmitter among the plurality of transmitters installed in the same room. By automatically controlling the variable loss circuits of other terminal devices to a high-loss state, the number of handsets in the on state can be substantially reduced, improving the S/N ratio and intelligibility of the transmitted signal. This has the remarkable effect of improving the

【図面の簡単な説明】[Brief explanation of drawings]

第1図はこの発明を実施した会議電話装置のシスデム構
成図、第2図は同じく主装置と端末装置の回路図、第3
図は同じく端末装置の制御回路の回路図である。 1・・・電話回線、2・・・主装置、3・・・接続ケー
ブル、4・・・端末装置、5・・・送話器、6・・・受
話器、10・・・受話信号線、11・・・送話信号線、
12・・・制御信号線、16・・・可変1Ω失回路、1
9・・・音声スイッチ回路、20・・・制御回路、22
・23・・・フリップフロップ。 第2図
FIG. 1 is a system configuration diagram of a conference telephone device embodying the present invention, FIG. 2 is a circuit diagram of the main device and terminal device, and FIG.
The figure is also a circuit diagram of the control circuit of the terminal device. DESCRIPTION OF SYMBOLS 1... Telephone line, 2... Main device, 3... Connection cable, 4... Terminal device, 5... Transmitter, 6... Receiver, 10... Receiving signal line, 11... Transmission signal line,
12... Control signal line, 16... Variable 1Ω loss circuit, 1
9... Audio switch circuit, 20... Control circuit, 22
・23...Flip-flop. Figure 2

Claims (1)

【特許請求の範囲】[Claims] 送話器が接続されるとともに共通の制御信号線に並列接
続された複数台の端末装置を使用する会議電話装置にお
いて、前記各端末装置は前記送話器の出力信号路に挿入
された可変損失回路と、前記送話器の出力信号レベルを
検出する音声スイッチ回路と、この音声スイッチ回路の
検出出力信号がセット信号として供給されるフリップフ
ロップとをそなえ、このフリップフロップの出力信号に
よって前記可変損失回路の損失量を制御するとともに前
記制御信号線に制御信号を供給しかつ前記フリップフロ
ップのリセット信号を供給するようにし、時間的に最も
早く前記音声スイッチ回路から検出出力信号がセット信
号として前記フリップフロップに供給されたときこのフ
リップフロップが属する端末装置の可変損失回路の損失
量だけを小さくすると同時に、その他の前記端末装置の
可変損失回路の損失量を前記制御信号線の制御信号によ
って小さくならないようにしている会議電話装置。
In a conference telephone system using a plurality of terminal devices connected in parallel to a common control signal line with a transmitter connected thereto, each of the terminal devices has a variable loss transmitter inserted into the output signal path of the transmitter. a voice switch circuit for detecting the output signal level of the transmitter, and a flip-flop to which the detection output signal of the voice switch circuit is supplied as a set signal, and the output signal of the flip-flop is used to detect the variable loss. In addition to controlling the loss amount of the circuit, a control signal is supplied to the control signal line and a reset signal of the flip-flop is supplied, so that the detection output signal from the audio switch circuit is sent to the flip-flop as a set signal at the earliest time. When the flip-flop is supplied to the control signal line, the loss amount of only the variable loss circuit of the terminal device to which this flip-flop belongs is reduced, and at the same time, the loss amount of the variable loss circuit of the other terminal devices is not reduced by the control signal of the control signal line. conference telephone equipment.
JP29439485A 1985-12-26 1985-12-26 Conference telephone equipment Pending JPS62154839A (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP29439485A JPS62154839A (en) 1985-12-26 1985-12-26 Conference telephone equipment

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP29439485A JPS62154839A (en) 1985-12-26 1985-12-26 Conference telephone equipment

Publications (1)

Publication Number Publication Date
JPS62154839A true JPS62154839A (en) 1987-07-09

Family

ID=17807164

Family Applications (1)

Application Number Title Priority Date Filing Date
JP29439485A Pending JPS62154839A (en) 1985-12-26 1985-12-26 Conference telephone equipment

Country Status (1)

Country Link
JP (1) JPS62154839A (en)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2009165770A (en) * 2008-01-21 2009-07-30 Fujishoji Co Ltd Game machine

Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS4857522A (en) * 1971-11-19 1973-08-13
JPS5469307A (en) * 1977-11-14 1979-06-04 Toa Electric Co Ltd Loudspeaking telephone set
JPS5776938A (en) * 1980-10-30 1982-05-14 Toa Tokushu Denki Kk Interphone

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JPS4857522A (en) * 1971-11-19 1973-08-13
JPS5469307A (en) * 1977-11-14 1979-06-04 Toa Electric Co Ltd Loudspeaking telephone set
JPS5776938A (en) * 1980-10-30 1982-05-14 Toa Tokushu Denki Kk Interphone

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
JP2009165770A (en) * 2008-01-21 2009-07-30 Fujishoji Co Ltd Game machine

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