JP3179641B2 - Loop type audio conference circuit - Google Patents

Loop type audio conference circuit

Info

Publication number
JP3179641B2
JP3179641B2 JP29025293A JP29025293A JP3179641B2 JP 3179641 B2 JP3179641 B2 JP 3179641B2 JP 29025293 A JP29025293 A JP 29025293A JP 29025293 A JP29025293 A JP 29025293A JP 3179641 B2 JP3179641 B2 JP 3179641B2
Authority
JP
Japan
Prior art keywords
transmission
signal
loop
circuit
loss
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 - Fee Related
Application number
JP29025293A
Other languages
Japanese (ja)
Other versions
JPH07143242A (en
Inventor
和人 広瀬
正明 滝沢
大一郎 高嶋
弘行 松井
靖浩 富田
勉 入島
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.)
Hitachi Ltd
Nippon Telegraph and Telephone Corp
Original Assignee
Hitachi Ltd
Nippon Telegraph and Telephone Corp
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 Hitachi Ltd, Nippon Telegraph and Telephone Corp filed Critical Hitachi Ltd
Priority to JP29025293A priority Critical patent/JP3179641B2/en
Publication of JPH07143242A publication Critical patent/JPH07143242A/en
Application granted granted Critical
Publication of JP3179641B2 publication Critical patent/JP3179641B2/en
Anticipated expiration legal-status Critical
Expired - Fee Related legal-status Critical Current

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  • Small-Scale Networks (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Telephonic Communication Services (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Description

【発明の詳細な説明】DETAILED DESCRIPTION OF THE INVENTION

【0001】[0001]

【産業上の利用分野】本発明は複数端末をル−プ状に接
続して多地点音声会議を行う場合の音声加算方式改良に
係る。
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an improvement in a voice addition method in a case where a plurality of terminals are connected in a loop to conduct a multipoint voice conference.

【0002】[0002]

【従来の技術】複数個の音声系端末(電話)を通信網を
介して結び音声会議を行う技術が知られている。特に端
末が2本の通話路(チャネル)を用い、それぞれの通話
路を異なった対地に接続することにより全体として環状
(ル−プ)伝送路を形成し、音声会議を行う技術も特開
平3−289850号公報(会議接続の確立方法及びそ
の装置)により公知である。上記公知例において、電話
会議を行う複数端末はネットワ−ク内に環状ノ−ドとし
て組み込まれ、各ノ−ドでは音声のドロップ/インサ−
トが行われる。更に詳しく言えば、ル−プからの受信音
声に対して過去に自端末が挿入した音声を差し引いてこ
れを自端末の受信音声とし、更に自端末の新しい音声を
加えてル−プへの送出信号とする、いわゆるN−1加算
方式が説明されている。
2. Description of the Related Art There is known a technology of connecting a plurality of voice terminals (telephones) via a communication network to hold a voice conference. In particular, Japanese Patent Laid-Open Publication No. Heisei 3 discloses a technology in which a terminal uses two communication paths (channels) and connects each communication path to a different ground to form a ring transmission loop as a whole and perform a voice conference. -289850 (method and apparatus for establishing a conference connection). In the above-mentioned known example, a plurality of terminals for performing a telephone conference are incorporated as ring nodes in a network, and each node has a voice drop / insertion function.
Is performed. More specifically, the voice received by the terminal in the past is subtracted from the received voice from the loop, and this is used as the received voice of the own terminal, and the new voice of the own terminal is added and transmitted to the loop. A so-called N-1 addition method using signals is described.

【0003】[0003]

【発明が解決しようとする課題】上記説明した技術にお
いては、ル−プ伝送路に対する送受信信号は帯域圧縮の
ため高能率符号化された信号が使用される。例えばパル
ス符号変調(PCM)されたμ−Law符号、あるいは
適応差分パルス符号変調(ADPCM)などである。こ
の様な符号はもちろん直接加減算ができないので直線符
号に変換してから加減算操作(N−1加算)を行わなけ
ればならない。このとき、いわゆる量子化雑音の混入が
さけられない。この量子化雑音はル−プ中の各ノ−ドで
発生してル−プを巡回する。ところが、ル−プ自体は信
号処理で言う完全積分器を構成しているものなので、ル
−プ内では雑音電力が増大し通話品質の劣化を招くとい
う現象が生じる。本発明の目的は上記したル−プ内雑音
を除去して良好な会議通話品質を維持しうる音声会議回
路を提供する事にある。
In the technique described above, a signal which has been encoded with high efficiency for band compression is used as a transmission / reception signal for a loop transmission line. For example, a pulse code modulated (PCM) μ-Law code or adaptive differential pulse code modulation (ADPCM) is used. Since such codes cannot be directly added or subtracted, they must be converted to linear codes and then subjected to an addition / subtraction operation (N-1 addition). At this time, mixing of so-called quantization noise cannot be avoided. This quantization noise is generated at each node in the loop and goes around the loop. However, since the loop itself constitutes a complete integrator in signal processing, a phenomenon occurs in the loop where noise power increases and communication quality deteriorates. SUMMARY OF THE INVENTION It is an object of the present invention to provide a voice conference circuit capable of maintaining good conference call quality by removing the above-mentioned loop noise.

【0004】[0004]

【課題を解決するための手段】ル−プ伝送路を介して音
声のN−1加算を実施せんとする場合、伝送路符号化お
よび復号化がなされる事から伝送路上で更に量子化雑音
が増大することがさけられない。本発明においては、図
1に原理構成図として示したように、伝送路符号化で発
生した量子化雑音が一回ル−プを周回するのは良しとし
て2回以上ル−プを巡回する事を防ぐ様構成した。即
ち、ル−プ型伝送路に接続されて、伝送路からの受信入
力信号に自身の送信入力信号を加算して伝送路への送信
出力信号とし、また伝送路からの受信入力信号から自身
の送信入力信号を除去して受信出力信号とする音声会議
用4端子回路網の構成であって、伝送路からの受信入力
符号を直線符号に復号化して受信信号とする手段、かか
る受信信号に減衰を付与して受信減衰信号とする手段、
上記受信減衰信号に送信入力端子からの入力信号を加算
し、かつ局部シミュレ−ト手段が作成する消去信号を減
算して送信信号を作成する手段、かかる送信信号を圧縮
して送信出力符号を得る伝送路符号化手段を含み、該伝
送路符号化手段において送信出力符号に含まれる前記送
信信号との差分である量子化雑音を抽出し、前記送信入
力信号とともに前記局部シミュレ−ト手段の入力とし、
かかる局部シミュレ−ト手段としては、ル−プ型伝送路
に沿うて一周する信号の伝送状況をル−プ遅延とル−プ
損失と伝送路符号化と復号化のくりかえしにて等価表現
となる様に構成したものである。
In the case where N-1 addition of voice is to be performed via a loop transmission path, quantization noise is further reduced on the transmission path because the transmission path is encoded and decoded. It cannot be increased. In the present invention, as shown in the principle configuration diagram in FIG. 1, it is good that the quantization noise generated in the transmission path coding goes around the loop once or more than twice. It was configured to prevent That is, it is connected to a loop-type transmission line, adds its own transmission input signal to the reception input signal from the transmission line to generate a transmission output signal to the transmission line, and outputs its own reception signal from the transmission line. A configuration of a four-terminal audio conferencing network for removing a transmission input signal and providing a reception output signal. Means for giving a reception attenuation signal by adding
A means for adding a signal input from a transmission input terminal to the reception attenuation signal and subtracting an erasure signal generated by the local simulating means to generate a transmission signal, and compressing the transmission signal to obtain a transmission output code. A transmission path encoding means for extracting quantization noise, which is a difference from the transmission signal included in the transmission output code, in the transmission path encoding means as an input to the local simulating means together with the transmission input signal; ,
As such local simulating means, the transmission status of a signal circulating along a loop-type transmission path is equivalently represented by loop delay, loop loss, and repeated transmission path coding and decoding. It is configured in such a manner.

【0005】[0005]

【作用】図1に示した様に、伝送路符号化手段にて余儀
なく発生した量子化雑音成分は局部シミュレ−ト手段へ
帰還される。局部シミュレ−ト手段は伝送ル−プ一周に
つきその伝送の状況を遅延要素、損失を表現する要素、
更には伝送路符号化・復号化手段等の要素により等価表
現したものである。従って送信入力信号とともに上記抽
出された量子化雑音成分は伝送路の一周と同様な過程を
経て消去信号として前記加減算手段へ到達するので、自
端末で発生した信号だけは自端末が責任をもって消去す
る事ができる。かくして信号(雑音を含む)はル−プを
再巡還しないので信号対雑音の比もしくは通話品質が著
しく向上するものである。
As shown in FIG. 1, the quantization noise component generated by the transmission path coding means is fed back to the local simulating means. The local simulating means includes a delay element and a loss element representing the transmission status per round of the transmission loop.
Furthermore, it is equivalently expressed by elements such as transmission path encoding / decoding means. Therefore, the extracted quantization noise component together with the transmission input signal reaches the adding / subtracting means as an erasure signal through a process similar to one round of the transmission path, so that only the signal generated by the own terminal is erased by the own terminal with responsibility. Can do things. Thus, the signal (including noise) does not return to the loop, thereby significantly improving the signal to noise ratio or speech quality.

【0006】[0006]

【実施例】以下、図面を用いて本発明の実施例を詳細に
説明する。まず、図4は本発明による音声会議回路を伝
送路を介してル−プ状に接続した状況を説明する。N個
の会議回路(1〜N)は通話路(CHi、CHj)を介
してリング接続され、受信入力信号Rin(t)と送信
出力信号Sout(t)を送受信する。また、送信入力
信号Sin(t)と受信出力信号Rout(t)をリン
グの外に入出力するものである。
Embodiments of the present invention will be described below in detail with reference to the drawings. First, FIG. 4 illustrates a situation where the audio conference circuit according to the present invention is connected in a loop through a transmission line. The N conference circuits (1 to N) are ring-connected via communication paths (CHi, CHj), and transmit and receive a reception input signal Rin (t) and a transmission output signal Sout (t). Further, the transmission input signal Sin (t) and the reception output signal Rout (t) are input / output outside the ring.

【0007】次に、図1を用いて本発明の音声会議回路
の原理構成を説明する。通話路からの受信音声は受信入
力端子30に与えられる。これは高能率符号化された圧
縮符号であるので伝送路復号化手段300によって直線
符号形式の受信信号に伸長され、更にル−プ損失符号化
手段500によって信号の減衰が付与される。信号減衰
はル−プ内の伝送誤りを自然消滅させるために本質的に
必要なものである。減衰された信号には加減算手段70
0により送信入力端子10からの送信入力信号と局部シ
ミュレ−ト手段600からの消去信号が加減算され送信
信号が作成される。送信信号は伝送路符号化手段400
によって高能率符号化され、該送信出力符号は送信出力
端子40から伝送路に送出される。ここで伝送路符号化
手段により送信信号に対して符号化雑音(量子化雑音)
が付加される。該雑音を抽出し、送信入力信号とともに
局部シミュレ−ト手段600に入力する。局部シミュレ
−ト手段は、本回路の送信出力40から受信入力30ま
でのル−プ一周伝送をル−プ遅延量、ル−プ減衰量、更
に符号化と復号化のくりかえし等の要素により正確にシ
ミュレ−トするものであり、本発明の重要な特徴であ
る。
Next, the principle configuration of the audio conference circuit of the present invention will be described with reference to FIG. The voice received from the communication path is given to the reception input terminal 30. Since this is a high-efficiency coded compression code, it is expanded into a received signal in the form of a linear code by the transmission path decoding means 300, and further attenuated by the loop loss coding means 500. Signal attenuation is essentially necessary to eliminate transmission errors in the loop naturally. Addition / subtraction means 70 is applied to the attenuated signal.
By means of 0, a transmission input signal from the transmission input terminal 10 and an erasure signal from the local simulating means 600 are added or subtracted to create a transmission signal. The transmission signal is transmitted to the transmission path coding unit 400.
The transmission output code is transmitted from the transmission output terminal 40 to the transmission line. Here, coding noise (quantization noise) is applied to the transmission signal by the transmission path coding means.
Is added. The noise is extracted and input to the local simulating means 600 together with the transmission input signal. The local simulating means is capable of accurately performing loop loop transmission from the transmission output 40 to the reception input 30 of the present circuit by means of loop delay, loop attenuation, and repeated coding and decoding. This is an important feature of the present invention.

【0008】本発明によれば、音声会議回路に入力され
る送信入力信号と余儀なく注入される量子化雑音は該消
去信号によって正確に消去され、1回だけのル−プ循環
で終わる。この再循環の阻止が通話品質(信号対雑音
比、無通話時雑音、エコ−除去)の向上に寄与するとこ
ろ大なのである。なお、受信出力信号は、例えば送信信
号からタップして、入力信号を差し引くことにより受信
出力端子20にて得ることが出来る。
According to the present invention, the transmission input signal input to the audio conferencing circuit and the quantization noise forced to be injected are accurately canceled by the cancellation signal, ending with only one loop circulation. The prevention of this recirculation contributes to the improvement of speech quality (signal-to-noise ratio, noise during no speech, and echo reduction). The reception output signal can be obtained at the reception output terminal 20 by, for example, tapping the transmission signal and subtracting the input signal.

【0009】図1の原理をより具体的に実施した例を図
2に示す。本例はル−プ中に付与すべき損失を各会議回
路に均一に分散させた場合であり、例えばA=−1db
前後の値を付与する。ル−プ全体では、N端末に対し−
NAdbの減衰となる。図2の中に示した伝送路復号化
回路300、ル−プ損失符号化回路500、加減算回路
700、ル−プシミュレ−ト回路600、伝送路復号化
回路400は、全て図1のそれらと対応している。また
ル−プ全体として、他の端末会議回路の一部が省略書き
されている。更に、図2の中の記号Cは8ビットμ−則
符号化回路、Dは8ビットμ−則復号化回路、Tはル−
プ全体の遅延量を集中表現した遅延メモリ、αは損失を
付与するためのα<1.0の乗算係数をあらわしてい
る。(上記のAに対してA=20Log(α)であ
る。)受信入力端子30に到達した信号は送信出力端子
40へ出力されるまでに等価的に1回だけα減衰、符号
化、復号化のプロセスを経由するので局部シミュレ−ト
回路は本プロセスをN回くりかえし、かつ集中遅延Tの
継続接続によって構成される。
FIG. 2 shows an example in which the principle of FIG. 1 is more specifically implemented. In this example, the loss to be provided during the loop is uniformly distributed to each conference circuit. For example, A = -1db
Assign values before and after. In the entire loop, N-
NAdb is attenuated. The transmission line decoding circuit 300, loop loss encoding circuit 500, addition / subtraction circuit 700, loop simulation circuit 600, and transmission line decoding circuit 400 shown in FIG. 2 all correspond to those in FIG. are doing. Further, a part of the other terminal conference circuit is abbreviated as the whole loop. Further, symbol C in FIG. 2 is an 8-bit μ-law encoding circuit, D is an 8-bit μ-law decoding circuit, and T is a rule.
.Alpha. Represents a multiplication coefficient of .alpha. <1.0 for giving a loss, in which the delay amount of the entire loop is intensively expressed. (A = 20 Log (α) with respect to A described above.) A signal reaching the reception input terminal 30 is equivalently α-attenuated, encoded, and decoded only once before being output to the transmission output terminal 40. Therefore, the local simulate circuit repeats this process N times and is constituted by the continuous connection of the concentrated delay T.

【0010】次に、ル−プに挿入する損失をある1個の
端末会議回路にのみ集中配置し(この損失を以下L=α
のN剰とする)、他N−1個の会議回路に対しては損失
を配置しない(α=1.0)の場合の具体的構成例を図
3に示す。回路に付与した記号番号は図1、図2と同様
であるから説明は省略する。ル−プ損失として損失集中
端末に損失Lが与えられ、他の端末ではα=1.0が与
えられている。自身を除いたル−プ一周の損失の見え方
が損失集中端末からみるとα=1.0、それ以外の端末
からみると常に損失Lと見えるので局部シミュレ−ト回
路内の損失表現が上記と逆転している。また同一信号レ
ベルの符号化と復号化は複数回くりかえしても同一の結
果となる(同一の符号語と同一の代表値のくりかえし)
ので符号化と復号化のくりかえしは1回だけで全体のシ
ミュレ−トとなっている。
Next, the loss to be inserted into the loop is concentrated on only one terminal conference circuit (this loss is hereinafter referred to as L = α).
FIG. 3 shows a specific configuration example in a case where no loss is allocated to the other N-1 conference circuits (α = 1.0). The symbol numbers assigned to the circuits are the same as those in FIGS. The loss L is given to the loss concentration terminal as the loop loss, and α = 1.0 is given to other terminals. The loss appearance in the loop around the loop excluding itself is α = 1.0 when viewed from the loss concentration terminal, and always L when viewed from the other terminals, so the loss expression in the local simulate circuit is described above. Is reversed. In addition, encoding and decoding at the same signal level produce the same result even when repeated a plurality of times (repeat of the same code word and the same representative value).
Therefore, the encoding and decoding are repeated only once to form the entire simulation.

【0011】図2、図3の具体例では伝送路符号化とし
て8ビットのμ−則コーデックを用いた場合を示した
が、これを適応差分パルス符号変調(ADPCM)コー
デックを用いてもよい。その場合、伝送路符号化回路で
発生する量子化雑音の抽出はADPCMコーデックの入
力信号と局部復号器出力の差分とすればよい。以上のよ
うに伝送路符号化部における量子化雑音を抽出して局部
シミュレ−ト回路にこれを加え、更に局部シミュレ−ト
回路としてはル−プ一周の伝送状態を等価的に正確表現
するという本発明を用いるならば、信号対雑音電力比、
無通話時雑音等の電気的性能を2〜4db向上する事が
できる。
In the specific examples shown in FIGS. 2 and 3, a case where an 8-bit μ-law codec is used as the transmission path coding is shown. However, an adaptive differential pulse code modulation (ADPCM) codec may be used. In this case, the extraction of the quantization noise generated in the transmission path coding circuit may be performed by using the difference between the input signal of the ADPCM codec and the output of the local decoder. As described above, the quantization noise in the transmission line coding unit is extracted and added to the local simulation circuit, and the local simulation circuit expresses the transmission state of the loop around equivalently and accurately. With the present invention, the signal to noise power ratio,
The electrical performance such as noise during non-communication can be improved by 2 to 4 db.

【0012】なお、以上説明した信号処理は近年発達の
著しいデジタル信号処理プロセッサ(DSP)にて容易
に実現される。DSPを用い、デジタル通信網(ISD
N)に接続される音声会議端末のハ−ドウェア構成を図
5に示す。同図に示した様にDSPは音声入出力系(マ
イク、スピ−カ)とPCMコーデックを介して、またI
SDN伝送路とISDNインタフェ−ス回路を介して接
続される。外部にはプログラムを格納するROM、また
デ−タを格納するRAMが接続され、装置全体は上位の
マイクロプロセッサMPUにて制御される。この様に会
議端末装置はDSPを用いて容易に構成される。
The above-described signal processing can be easily realized by a digital signal processor (DSP), which has been remarkably developed in recent years. Digital communication network (ISD) using DSP
FIG. 5 shows the hardware configuration of the audio conference terminal connected to N). As shown in the figure, the DSP is connected to a voice input / output system (microphone, speaker) and a PCM codec.
It is connected to an SDN transmission line via an ISDN interface circuit. A ROM for storing programs and a RAM for storing data are connected to the outside, and the entire apparatus is controlled by a higher-order microprocessor MPU. As described above, the conference terminal device is easily configured using the DSP.

【0013】[0013]

【発明の効果】以上説明した様に本発明のル−プ型音声
会議回路によれば、伝送路符号化部で発生する量子化雑
音を抽出してル−プ一周の伝送をシミュレ−トする局部
シミュレ−ト回路にこれを加え消去信号を作成する手段
を講じたので量子化雑音のル−プ巡還を阻止する事が出
来、良好な通話品質を確保する事が可能である。通話品
質向上の一例を定量的に述べると、N=5地点をル−プ
接続し伝送路符号化手段としてμ−則コーデック(8ビ
ット)を用いた場合、ル−プ損失の挿入の仕方と測定場
所が信号挿入地点から第何地点であるかにも依存する
が、2〜5db程度の効果を得ることが出来る。伝送路
符号化手段として7ビットADPCMコーデックを用い
た場合、やはり3〜4dbの信号対雑音比(S/N)向
上をみることができる。もちろん無通話時雑音も3〜5
db向上する。更にS/Nの向上は同時に自身の送話信
号の自身への受信出力消去能力の向上にも寄与し、本発
明の効果は大きい。
As described above, according to the loop type voice conference circuit of the present invention, the quantization noise generated in the transmission path coding unit is extracted to simulate the transmission around the loop. Since a means for creating an erasure signal by adding this to the local simulate circuit is provided, loopback of the quantization noise can be prevented, and good speech quality can be ensured. Quantitatively describing an example of the improvement of speech quality, when N = 5 points are connected in a loop and a μ-law codec (8 bits) is used as a transmission path coding means, how to insert a loop loss and An effect of about 2 to 5 db can be obtained although it depends on the number of measurement points from the signal insertion point. When a 7-bit ADPCM codec is used as the transmission path coding means, the signal-to-noise ratio (S / N) can be improved by 3 to 4 db. Of course, the noise during no call is 3-5.
db is improved. Further, the improvement in S / N also contributes to the improvement of the ability to cancel the reception output of the own transmission signal to itself, and the effect of the present invention is great.

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

【図1】本発明の原理構成図。FIG. 1 is a diagram showing the principle configuration of the present invention.

【図2】本発明の会議回路装置の回路構成図(損失:各
端末に分散)。
FIG. 2 is a circuit diagram of a conference circuit device of the present invention (loss: distributed to each terminal).

【図3】本発明の会議回路装置の別の回路構成図(損
失:一端末に集中)。
FIG. 3 is another circuit configuration diagram (loss: concentrated on one terminal) of the conference circuit device of the present invention.

【図4】本発明の会議回路装置をリング状に接続したル
−プ型音声会議回路の構成図。
FIG. 4 is a configuration diagram of a loop type audio conference circuit in which the conference circuit devices of the present invention are connected in a ring shape.

【図5】本発明の会議回路装置の具体的ハ−ドウェア構
成例図。
FIG. 5 is a diagram showing an example of a specific hardware configuration of the conference circuit device of the present invention.

【符号の説明】[Explanation of symbols]

1、2、〜、N…音声会議回路、 10………………送信入力端子、 20………………受信出力端子、 30………………受信入力端子、 40………………送信出力端子、 300、400…直線符号と高能率符号を相互変換する
回路、 700……………加減算回路、 500……………ル−プ損失付与符号化回路、 600……………局部シミュレ−ト回路。
1, 2, ..., N: audio conference circuit, 10: transmission input terminal, 20: reception output terminal, 30: reception input terminal, 40: reception input terminal ..., Transmission output terminals, 300, 400, a circuit for mutually converting between a linear code and a high-efficiency code, 700,..., An addition / subtraction circuit, 500,. ...... Local simulation circuit.

───────────────────────────────────────────────────── フロントページの続き (72)発明者 高嶋 大一郎 神奈川県横浜市戸塚区戸塚町216番地株 式会社日立製作所情報通信事業部内 (72)発明者 松井 弘行 東京都千代田区内幸町一丁目1番6号日 本電信電話株式会社内 (72)発明者 富田 靖浩 東京都千代田区内幸町一丁目1番6号日 本電信電話株式会社内 (72)発明者 入島 勉 東京都千代田区内幸町一丁目1番6号日 本電信電話株式会社内 (56)参考文献 特開 平3−289850(JP,A) 特開 平5−129989(JP,A) 特開 平2−148926(JP,A) (58)調査した分野(Int.Cl.7,DB名) H04M 3/42 - 3/58 H04B 14/00 - 14/08 H04B 1/76 - 3/60 H04B 7/005 - 7/015 ──────────────────────────────────────────────────続 き Continuing on the front page (72) Inventor Daiichiro Takashima 216 Totsuka-cho, Totsuka-ku, Yokohama-shi, Kanagawa Prefecture Inside the Hitachi, Ltd. Information and Communications Division (72) Inventor Hiroyuki Matsui 1-1-1, Uchisaiwaicho, Chiyoda-ku, Tokyo No. 6 Inside the Telegraph and Telephone Corporation (72) Inventor Yasuhiro Tomita 1-1-6 Uchisaiwaicho, Chiyoda-ku, Tokyo Japan Within (72) Inventor Tsutomu Irishima 1-1-1 Uchisaiwaicho, Chiyoda-ku, Tokyo No. 6 Inside the Telegraph and Telephone Corporation (56) References JP-A-3-289850 (JP, A) JP-A-5-129989 (JP, A) JP-A-2-148926 (JP, A) (58) Field surveyed (Int.Cl. 7 , DB name) H04M 3/42-3/58 H04B 14/00-14/08 H04B 1/76-3/60 H04B 7/005-7/015

Claims (1)

(57)【特許請求の範囲】(57) [Claims] 【請求項1】伝送路上にループ型に接続された複数の音
声会議回路同士で会議通信を行う音声会議回路であっ
て、上記伝送路からの受信入力符号を復号化して受信入
力信号を得る復号化手段と、上記受信入力信号に対し上
記ループを周回する際に発生した損失を軽減する処理を
行うループ損失処理手段と、上記ループ損失を軽減した
受信入力信号に自分自身の音声会議回路において入力さ
れた送信入力信号を加算し、かつループ周回後の自分自
身の音声会議回路の送信入力信号をシミュレートする局
部シミュレート手段において作成した消去信号を減算し
て送信信号を得る加減算手段と、上記送信信号を符号化
して送信出力符号を得て上記伝送路へ送信し、かつその
送信出力符号に含まれる上記送信信号との差分である量
子化雑音を抽出する符号化手段とを有し、上記局部シミ
ュレート手段には上記送信入力信号と上記量子化雑音を
入力し、上記送信信号から自分自身の音声会議回路にお
いて入力された送信入力信号を減算した信号を音声会議
回路からの受信出力信号として出力することを特徴とす
るループ型音声会議回路。
1. A plurality of sounds connected in a loop on a transmission line.
A voice conference circuit that performs conference communication between voice conference circuits.
To decode the input code received from the transmission path
Decoding means for obtaining a force signal;
Processing to reduce the loss that occurs when going around the loop
Loop loss processing means for reducing the loop loss
Enter the received input signal in your own audio conference circuit.
The added transmission input signal and the self
A station that simulates the transmission input signal of a live audio conference circuit
Subtracts the erasure signal created by the
Adding / subtracting means for obtaining a transmission signal by encoding the transmission signal
To obtain a transmission output code and transmit it to the transmission path, and
Amount that is the difference from the above transmission signal included in the transmission output code
Encoding means for extracting sub-sampling noise;
The transmission input signal and the quantization noise
Input from the above transmission signal to your own audio conference circuit.
Audio conferencing by subtracting the transmitted input signal
Output as a reception output signal from the circuit.
Loop type audio conference circuit.
JP29025293A 1993-11-19 1993-11-19 Loop type audio conference circuit Expired - Fee Related JP3179641B2 (en)

Priority Applications (1)

Application Number Priority Date Filing Date Title
JP29025293A JP3179641B2 (en) 1993-11-19 1993-11-19 Loop type audio conference circuit

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
JP29025293A JP3179641B2 (en) 1993-11-19 1993-11-19 Loop type audio conference circuit

Publications (2)

Publication Number Publication Date
JPH07143242A JPH07143242A (en) 1995-06-02
JP3179641B2 true JP3179641B2 (en) 2001-06-25

Family

ID=17753732

Family Applications (1)

Application Number Title Priority Date Filing Date
JP29025293A Expired - Fee Related JP3179641B2 (en) 1993-11-19 1993-11-19 Loop type audio conference circuit

Country Status (1)

Country Link
JP (1) JP3179641B2 (en)

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR102145121B1 (en) * 2017-01-13 2020-08-14 가부시키가이샤 에버트론 Fryer

Families Citing this family (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR101350599B1 (en) * 2007-04-24 2014-01-13 삼성전자주식회사 Method and apparatus for Transmitting and Receiving Voice Packet

Cited By (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
KR102145121B1 (en) * 2017-01-13 2020-08-14 가부시키가이샤 에버트론 Fryer

Also Published As

Publication number Publication date
JPH07143242A (en) 1995-06-02

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