JPH04352557A - Telephone conference device - Google Patents

Abstract

PURPOSE:To constitute a telephone conference to be easily used under an ATM communication network by digitally adding subtracting voice signals and forming a voice bridge function by a voice buffer completing transmission for the delay time of one loop. CONSTITUTION:The respective telephone conference devices on the loop are housed in the ATM network and logically connected by an ATM line 3-8. The in-network delay fluctuation of a voice cell from the ATM network is absorbed by an absorbing mechanism 3-7 and transmitted to subtracter 3-5 at fixed speed. The subtracter 3-5 obtains information, which is added and synthesized in the preceding cycle by the own telephone conference device, from a buffer 3-6 completing the transmission and generates the voice cell extracting only the voice information of the other telephone conference device by subtracting and decomposing the information from the information of the voice cell. Voice information adding and synthesizing the voice of the other telephone conference device is reproduced by a D/A converter 3-1. On the other hand, voice information in a cell buffer 3-3 is transmitted to the ATM network after being added and synthesized to the voice cell, which is generated by the subtracter 3-5, of the other telephone conference device by an adder 3-4.

Description

[Detailed description of the invention]

0001

[Industrial Field of Application] The present invention is a teleconferencing device, that is, by using a plurality of telephones and a packet communication network, a local area network (LAN), or an ATM switching network, users of the aforementioned plurality of telephones can communicate with each other. The present invention relates to a teleconference device that can be suitably used to implement a teleconference.

0002

[Prior Art] When a plurality of people (N people: members of a conference call) who are geographically distant from each other hold a conference call,
Additionally synthesizes the voices of m speakers (m = 1 in normal usage mode) and distributes the synthesized voices to N-m people,
A so-called audio bridge device is required. When implementing a conference call, how the audio bridge device is placed within the communication network or at the telephone terminal greatly influences the economic efficiency and convenience of the conference call service. As for the arrangement method of the audio bridge device, for example, NTT Research Practical Application Report Vol. 34 No. 4 "Voice Conference Communication System" (1985)
As disclosed in 2010), an intra-network voice bridging method and a terminal voice bridging method are known. The in-network voice bridge method is a method in which one or more voice bridge functions are installed in a network such as a switching network, separate from the terminal or telephone, and one voice bridge is usually shared by multiple users. . On the other hand, the terminal voice bridge method is a method in which a voice bridge function is installed in each telephone. The former intra-network voice bridging method requires the incorporation of a voice bridging function within the communication network, which requires adding or changing functions to the communication network, whereas the latter terminal voice bridging method requires this. Such additions and changes are not necessary and can be realized using only the functions of the terminal's audio addition and distribution circuit. Therefore, when using the terminal voice bridge method, it has excellent flexibility and mobility, and can easily implement teleconferences not only on public networks but also on private networks for companies. In the past, the mainstream method for realizing teleconferences was the intra-network bridging method, but in recent years, advances in digital signal processing LSIs such as DSPs have made it easier to realize voice bridging at low cost. This has made it possible to put the terminal bridge method into practical use.

0003

[Problem to be Solved by the Invention] As a specific method of the terminal bridge method mentioned above, which is being put into practical use as a method for realizing telephone conferences, there is a method using a meeting phone (
(hereinafter referred to as the "meeting phone method"). This method uses ISDN basic interface 2B+D, and for example, "Voice-related applications using ISDN" (NTT Technology Journal, vol.
．． 2, No. 10, 1990). According to this, the meeting phone system consists of a three-way communication circuit consisting of two B channels (B1 channel, B2 channel) and three audio adders as an audio bridge function.
It has an encoder (COD) and a decoder (DEC) for transmitting and receiving audio signals for each B channel (see FIG. 5). Furthermore, as a specific example, FIG. 6 shows the flow of transmitting and receiving audio signals when a four-party conference is held. In the configuration shown in FIG. 6, telephone B receives a voice signal (a) from telephone A.
, and adds its own transmission signal (b) with the voice adder, and transmits it to telephone C. On the other hand, it receives the voice signal (c+d) from telephone C, adds its own transmission signal (b), and transmits it to telephone A. When each terminal connected in series performs this operation, the transmission signal from telephone B is transmitted to all other telephones. Furthermore, by adding the audio signals (a) and (c+d) received from both channels, it is possible to listen to the transmission signals of all other telephones. As described above, by using two channels for each telephone, it becomes possible to implement a conference call using a plurality of telephones. However, the above-mentioned method has the following problems. That is, (1) Each telephone requires two communication channels, and a normal telephone having only one communication channel cannot be used. (2) There is a large call delay. In other words, in the example shown in FIG. 6, telephone A must pass through the link (communication network) six times before it receives a response from telephone D to its own outgoing call.The latter will be explained in more detail as follows. , as follows. In the meeting phone system, generally when a conference is formed using N telephones, 2 (N-1) links are used. The transmission of call information in a conference call through multiple links is not a practical problem with conventional analog links with low latency or synchronous transfer mode (STM) links, but the number of links per link is not a problem. When using an ATM link that is expected to be 10 ms,
This is a serious problem. For example, assuming that the delay per ATM link is 30 milliseconds and the number of conference call members N is 11, the response delay will be 600 milliseconds based on the above relationship. A delay of 600 milliseconds is the upper limit for a natural conversation, and in this configuration example, it is inappropriate to have more than 11 members. When configuring a conference call using ATM links, it is clearly preferable to use a method that uses fewer links. The present invention has been made in view of the above circumstances, and its purpose is to solve the above-mentioned problems in the conventional technology, and to provide an easy-to-use and economical telephone conference system in an ATM communication network environment. An object of the present invention is to provide a teleconference device that allows users to configure a telephone conference.

0004

[Means for Solving the Problems] The above-mentioned object of the present invention is to provide a system in which a telephone conference is realized by connecting a plurality of telephone conference devices in a loop through one communication channel, from a voice block received from a receiving line. loop 1
means for arithmetically subtracting the linear PCM audio code that has been arithmetic added in the previous cycle; means for arithmetically adding the newly generated linear PCM audio code to the audio block and transmitting it as a new audio block; means for holding the PCM voice code for a time equal to or longer than the time it takes for the voice block to make one round of the loop; By arithmetically subtracting the voice code that has been added, the sum of the voice codes of other terminals is extracted, and the newly generated voice code is arithmetically added to the voice block from which the voice code from one round ago has been subtracted, and the voice block is This is achieved by a teleconferencing device characterized in that it transmits audio information to all other terminals by transmitting as .

[0005]

[Operation] In the teleconference device according to the present invention, each telephone that makes up the teleconference has means for holding the transmitted audio signal at least for the time required for the audio signal to go around the loop once (transmitted audio buffer). When each telephone receives one voice block from the loop, it subtracts its own voice signal added one cycle before from that block, uses this as the received voice of its own telephone, and also sends the new voice signal of its own telephone. is added to the above signal and transmitted. Note that at this time, the own audio signal is stored in the above-mentioned transmitted audio buffer and used as subtraction information in the next cycle. As described above, the present invention is characterized in that it performs addition and subtraction of audio signals in digital format and realizes the audio bridge function by using a transmitted audio buffer with a delay time of one round of the loop. That is, in an electronic conference, the voices of speakers are arithmetically added and transmitted to a plurality of listeners. If this addition is performed using digital audio signals, subtracting the added signal Y from the audio signal Z after addition will result in the original signal X before addition (X=Z−
Y) can be obtained correctly. On the other hand, when this calculation is performed on an analog audio signal, noise and distortion are accumulated each time the signal is transmitted or processed, so even if the above calculation Z-Y is performed, the original signal before addition is X cannot be obtained in general.

[0006]

Embodiments Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings. FIG. 1 is a diagram showing the configuration and processing flow of a teleconference device that is an embodiment of the present invention. In the figure, 3-1 is a D/A converter that converts audio cells into linear PCM signals and further converts them into audio, and 3-2 is a D/A converter that converts audio information to be transmitted into linear PCM signals, and further converts audio cells into audio cells. An A/D converter 3-3 converts the A/D converter 3-2 into
a cell buffer for holding the voice cell generated in 3 until the voice cell to which the information part contents of the voice cell should be added arrives;
-6 is a transmitted audio buffer that holds the contents of the cell transmitted in the cell buffer 3-3, that is, the audio code;
3-7 is a delay fluctuation absorption mechanism that temporarily stores voice cells received from the ATM network, absorbs delay fluctuations occurring within the network, and transmits the cells at regular intervals; 3-5 is the delay fluctuation absorption mechanism 3; - a subtracter for subtracting the content of the information part of the voice cell in the transmitted voice buffer 3-6 corresponding to this voice cell from the content of the information part of the voice cell output from 7;
3-4 is an adder that adds the voice cell output from the subtracter 3-5 and the information part of the voice cell to be transmitted next, which has been accumulated in the cell buffer 3-3, and transmits it to the ATM network. , 3-8 indicate an ATM line that accommodates a reception channel for receiving ATM cells from other conference terminals and a transmission channel for transmitting ATM cells to other ATM terminals.

[0007] The operation of this embodiment will be described below for a case where four telephone conference devices A, B, C, and D are used. If there are x cells on the loop, the system operates in x periods. Each teleconference device is accommodated in the ATM network by ATM lines 3-8, and is logically connected on a loop by logical lines using ATM cells. Intra-network delay fluctuations of voice cells received from the ATM network are absorbed by the delay fluctuation absorbing mechanism 3-7, and sent to the subtracter 3-5 at a constant speed. Inside this voice cell,
There is audio information that has been added and synthesized by each teleconference device including the own teleconference device. The subtracter 3-5 obtains the information added and synthesized by the own telephone conference device in the previous cycle from the already-sent buffer 3-6, and subtracts and decomposes the information from the information of the voice cell to obtain the voice of the other conference phone device. Generate a voice cell from which only information is extracted. Then, the D/A converter 3-1 reproduces audio information obtained by adding and synthesizing the voices of other teleconference devices. On the other hand, the audio information to be transmitted accumulated in the cell buffer 3-3 is processed by the adder 3-4, and the audio information generated by the subtracter 3-5 is converted into audio cells containing only the audio information of other teleconference devices. The signals are added and combined and transmitted to the ATM network. By each conference call device performing the above operations on a loop,
Each teleconference device can always reproduce audio information that has been additively synthesized by other teleconference devices, and conference-type communication can be realized between a plurality of teleconference devices.

FIG. 2 is a diagram showing a situation in which four teleconference devices A, B, C, and D are connected in a loop to conduct an audio conference. Each teleconference device has one uplink and one downlink
They are connected in a loop through an ATM communication network or the like using links. On this loop, an audio cell having content obtained by adding and synthesizing the audio information of each teleconferencing device circulates, and each teleconferencing device subtracts the information of its own teleconferencing device from this audio cell, and then The information is played back and a new voice cell is generated and sent to the next conference call device. As a result of the above operations, the audio information of one teleconferencing device is transmitted to all other teleconferencing devices without distortion, and the audio information of all other teleconferencing devices is added and synthesized in that teleconferencing device. Conference-type communication can be realized by obtaining voice that is unique to the user. FIG. 3 shows a cell configuration in which the timing of subtraction and addition is controlled according to the number of cells on the loop in the above embodiment, and N telephone conference devices (terminals).
FIG. 2 is a diagram illustrating cases 1 to N. In the figure, 5-1 is a header for transferring voice cells on the communication network. 5-2
1 to 5-2N indicate conference headers that control the timing for subtracting and adding audio from terminal 1 to terminal N. Further, 5-3 is audio information. When conference terminal i adds its own terminal information and transmits a voice cell, it writes a serial number in the conference header assigned to it. This serial number is a circulating positive number whose maximum value is greater than the number x of voice cells that can exist in the loop. The conference terminal stores the serial number written in the conference header together with the audio information in the aforementioned sent buffer. Then, by subtracting the audio information in the transmitted buffer corresponding to the contents of the conference header assigned to the own terminal from the received audio cells, the audio information added in the previous cycle is deleted from the loop. becomes possible. At this time, by setting the maximum value of the serial number to a value greater than or equal to x, there is always only one cell in the transmitted buffer having the number corresponding to the received cell.

FIG. 4 is a diagram illustrating a compensation method when a voice cell is lost on a communication network. Loss of a voice cell occurs when there is no cell to be sent at the timing to send a voice cell in the buffer that absorbs delay fluctuations, or when the content of the conference header assigned to the received voice cell is invalid. Detected when they are continuous. In this case, the subtractor sets the value "0" to notify the cell discard in the conference header of terminals other than the own terminal, and the cell contents are "0".
A dummy cell written with "0" is sent out. The adder performs normal addition to this cell, sets a normal serial number in the conference header of its own terminal, and transmits the voice cell. On the other hand, a terminal that detects a conference header that notifies cell discard only adds audio information to be sent next without subtracting it, and sets a normal sequential number in its own conference header. Send voice cells. With the above operation, even if cell discard occurs, it is detected by the terminal on the receiving side of the link where cell discard occurs, and each terminal transmits cells at regular intervals regardless of whether cell discard is detected. , the situation where multiple terminals newly generate cells by discarding one cell is avoided. Although voice information is lost, the effect disappears in the time it takes for the cell to go around the loop once.

0010

[Effects of the Invention] As described in detail above, according to the present invention, a teleconference device is realized that is easy to use and allows economical teleconferencing in an ATM communication network environment. This has the remarkable effect that it can be done.

[0011]

[Brief explanation of drawings]

FIG. 1 is a diagram showing the configuration and processing flow of a teleconference device according to an embodiment of the present invention.

FIG. 2 is a diagram showing a communication state when a four-party conference is held using the telephone conference device of the embodiment.

FIG. 3 is a diagram showing the configuration of a voice cell according to an embodiment.

FIG. 4 is an explanatory diagram of a cell loss compensation method.

FIG. 5 is a diagram showing a configuration example of a three-way communication circuit of a currently used meeting phone.

FIG. 6 is a diagram showing communication status when a terminal bridge type four-party conference is held using the meeting phone shown in FIG. 5;

[Explanation of symbols]

3-1: D/A converter, 3-2: A/D converter, 3-3
: cell buffer, 3-4: adder, 3-5: subtracter, 3
-6: Transmitted voice buffer, 3-7: Delay fluctuation absorption mechanism, 3-8: ATM line, 5-1: Header, 5-2
1 to 5-2N: Conference header, 5-3: Audio information.

Claims (1)

[Claims] Claim 1: In a method for realizing a teleconference by connecting a plurality of teleconferencing devices (telephones) in a loop through one communication channel, arithmetic addition is performed from the audio block received from the receiving line one round before the loop. means for arithmetically subtracting the linear PCM audio code that has been generated; means for arithmetically adding a newly generated linear PCM audio code to the audio block and transmitting it as a new audio block; a voice code that is arithmetic added from the voice block received from the reception line one cycle before the loop stored in the voice code holding means; By arithmetic subtracting the sum of the voice codes of other terminals, the newly generated voice code is divided into 1
A teleconference device characterized in that audio information is transmitted to all other terminals by arithmetically adding a previous audio code to a subtracted audio block and transmitting the result as an audio block.