JPS6315559A - Electronic conference system - Google Patents

Abstract

PURPOSE:To attain efficient data transmission by selecting an encoding form or a decoding form in response to the traffic quantity of a communication line. CONSTITUTION:A conference support section 991 measures the traffic quantity of a communication line based on a signal from a talker identification device 510, a state detector 800 and an operation input device 850 or the like, sends a selection signal (VRUL) to a voice input device 400 and a voice output device 450 while taking other data quantity into account, and the transmission speed of voice data from the voice input device 400 is selected as required such as 64kpbs-84kpbs by means of the mu-LAWPCM, ADPCM or APCAB. Thus, the efficient data transmission is attained.

Description

[Detailed Description of the Invention] [Object of the Invention] (Industrial Application Field) The present invention provides an audio input device and an audio output device that change the encoding format and the decoding format according to the amount of traffic on a communication line. The present invention relates to an electronic conference system.

(Prior Art) As a conventional electronic conference system, the one shown in FIG. 7, for example, has been considered. As shown in the figure, this electronic conference system includes a microphone 1, a speaker 2, a television camera 3 for capturing an image of the conference room, a television camera 4 for capturing a manuscript, a television monitor 5, a facsimile device 6, and a hard copy device as a printer. 7. Said microphone 1, speaker 2
an audio unit 8 that encodes and decodes audio signals; an image unit 9 that encodes and decodes image signals; a control unit 10 that controls the image units; an operation unit 11 that inputs instructions for the control unit 9; It is composed of a transmission unit 12 that transmits and receives signals and image signals.

(Problem to be Solved by the Invention) However, in such an electronic conference system, the audio unit 7 performs data compression such as 16k APCAB.
Communication was carried out using kbps lines, but for example, when not transmitting video data or only transmitting audio, there was a problem that the capacity of the line was left over and efficient data transmission could not be performed. there were.

The present invention has been made to solve these conventional problems, and an object of the present invention is to provide an electronic conference system that can perform efficient data transmission.

[Structure of the Invention] (Means for Solving Problems) The present invention provides an input system for inputting drawing data, still image data, moving image data, and audio data; A conference unit system consisting of an output system that outputs data, a transmission system that controls the transmission of these data between remote points, and a control system that controls the input system, output system, and transmission system is connected via a line. In an electronic conference system in which a plurality of devices are connected, a means for measuring the traffic volume of a communication line, an audio input device for encoding voices uttered by conference attendees in an encoding format according to the measurement results, and the measurement results The present invention is characterized by comprising an audio output device that decodes audio data in a decoding format corresponding to the audio data and then outputs the decoded audio data.

(Operation) In the present invention, the traffic of a communication line is measured, the audio input device selects an encoding format according to the measurement result to encode the audio, and the audio output device encodes the audio in a decoding format according to the measurement result. is selected and the audio data is decoded.

゛ (Example) Hereinafter, an example of the present invention will be described in detail based on the drawings.

FIG. 1 is a configuration block diagram showing a conference unit system according to an embodiment of the electronic conference system of the present invention.

The control device 900 is connected to drawing, still image, moving image, and audio input devices, and to the control device 900 is connected to soft copy, hard copy, and audio output devices. Signals from these input devices and signals to the output device are subjected to processing such as selection and multiplexing according to several control signals, and are then transmitted.

Input devices include a plurality of drawing input devices 100 for inputting handwritten characters and figures, a still image input device 200 for inputting documents before or during the meeting, and images of conference attendees and the background of the conference hall. A moving image input device 300 for inputting audio, and an audio input device 400 for inputting audio are connected.

In addition, the output devices include a plurality of flat display devices 150 and a graphic display device (1) 160 that display input information of the own terminal and transmitted drawing type and still image type, and a graphic display bag @ (1) 1
A hard copy device 250 for making a copy of 60, a graphic display device (2>350) for displaying the input information of the own terminal and the transmitted video, and an audio input device 450 for outputting the transmitted audio input information are connected. has been done.

Roughly, the storage device 600 continuously records this input information. The transmission device 700 is connected to two other remote conference unit systems.

The above input/output information is transmitted through signals from devices such as an operation input device 850, a state detection device 800 that detects whether conference attendees are sitting or leaving their seats, and a speaker identification device 510 that determines the speaker based on the voices of the conversation participants. The tip is controlled.

Next, the voice input device 400 will be explained.

The input device @400 converts the voices of conference attendees into electrical signals and incorporates them into the electronic conference system, and FIG. 2 shows an embodiment thereof.

As shown in the figure, this audio input device 400 includes four microphones 402-1 to 402-4 arranged to collect the voices of all attendees, and four microphones 402-1 to 402-4 placed in a direction different from that of the conference attendees. A microphone 402-5 arranged with directivity, a microphone amplifier (MAMP) 403 that increases the output signals of the microphones 402-1 to 402-5, and a total noise canceller (NCAN>404 that removes room noise). , a dereverberation unit (ECAN) 405 that removes reverberation, and a variable gain unit (A
GC>406, a mixer unit (VMIX) 407 that synthesizes the output signals of each microphone, and a voice encoding unit (VCOD>408 that compresses and encodes audio).

These four microphones 402-1 to 402-4 are arranged with proper directionality taken into consideration so as to be able to collect the voices of all the attendees, and pick up the voices uttered by the conference attendees. Microphone 402-5 also picks up background noise signals.

The output signals from the microphone amplifier (MAMP) 403 and each microbon are amplified to an appropriate level.

The room noise removal unit 404 is a circuit that removes background noise that always exists in the conference room, and uses a background noise signal picked up by the microphone 402-5 whose directivity is directed in a direction different from that of the conference attendees as a reference signal. A noise canceller generates signals VB-1 to VB-5 from which background noise has been removed from microphone voices VA-1 to ■A-4.

The reverberation removal unit 405 has a function to prevent howling from occurring due to echoes generated by the sound output from the speakers reflecting off the walls and windows of the conference room, and from deteriorating call quality due to reverberation. Therefore, the left and right speaker signals j%-3
It inputs R and SL and generates signals @VC1 to VC-4 without reverberation using a two-channel echo quillense.

The variable gain unit (AGC) 406 uses the speaker identification code TID given by the speaker identification device 510 to select one of the four input signals VC-1 to VC-4 from a microphone whose directivity is directed toward the currently speaking speaker. The mixer section (
VMIX) 407 adds the output signal of the variable gain section 406.

The audio encoding unit (VCOD>408 compresses and encodes the output signal V of the mixer unit 407 according to the transmission rate using the encoding algorithm selection signal QVRUL sent from the control device 900, and further encodes the output signal V from the mixer unit 407 using an identifier. is added and output to the control device 900.

FIG. 3 is a circuit diagram of this audio encoding section 408.

As shown in the figure, the audio encoding unit 408 includes a compression algorithm storage unit 408-1 in which a compression algorithm program is stored, a selection switch 40B-2 for selecting a compression algorithm, and a compression algorithm that selects a compression algorithm for audio data. Encoding unit 408-3 compresses and encodes using an algorithm
and an identifier adding section 408-4 that adds control information.

This compression algorithm storage unit 408-1 has 64 KA
DPCM encoding program 40B-1a.

32k ADPDM encoding program 408-1b, 16
kAPCAB encoding program 408-1c.

and 8kAPCAB encoding program 408-1d are stored.

First, the selection switch 408-2 selects one encoding program in the compression algorithm storage section 408-1 according to the selection signal VRUL sent from the control device 900.

Next, the encoding unit 408-3 selects the jn selection switch 408
The output signal @V of the mixer section 407 is encoded by the encoding program selected by -2.

The identifier adding section 408-4 adds a transmitter identifier (TID) and a device address (SAD) to the encoded signal and outputs it to the control device 900 as a transmission signal VTX.

Next, the audio output device 450 will be explained.

The audio output device 450 converts an electrical signal into a high-pitched signal and outputs it through a speaker, and as shown in FIG. 4, it includes an identifier detection unit 451 that identifies control information, and an audio decoding unit that decodes audio data. (VDEC) 452 and a stereo audio generation unit (ST) that outputs stereo audio with delay.
GEN) 453, a right amplifier (R△MP>454, and a right speaker 455 that amplifies and emits the right audio), a left amplifier (LAMP) 456 that amplifies and emits the right audio, and a left speaker 457. .

The identifier detection unit 451 identifies the audio frame from the frame identifier FID, and identifies the partner terminal from the device address SAD.

FIG. 5 shows the configuration of the voice decoding section (VDEC) 452, and as shown in the figure, the voice decoding section 452
A compression algorithm storage unit 458 that stores a decoding program, a selection switch 459 that selects a decoding program, and a frame decomposition unit 460 that decomposes frames.
, a decoding unit 461 that decodes audio data using a selected algorithm, a RAMI 462 and a RAMI 463 in which decoded data corresponding to the terminal is stored, and a switch 464.

This compression algorithm storage unit 458 stores 64KADP.
CM decoding program 458-1.32 KADPCH decoding program 458-2.16 KAPCAB decoding program 458-3.8 KAPCAB decoding program 458-4 are stored.

Further, the selection switch 459 selects one decoding program in the compression algorithm storage section 458 in accordance with the selection signal VRUL sent by the control device 900.

RAMI462 and RAMI[463 store data used for decoding depending on the partner terminal.

First, the frame decomposition unit 460 decomposes the frame, extracts the partner terminal identification data CR, and outputs this to the switch 464.

Next, the switch 464 switches to R based on the partner terminal identification data CR.
Either the AMI 462 or the RAMlI 463 is selected and data used for decoding is output to the decoding section 461.

The rough decoding section 461 performs decoding using the compression algorithm (ie, decoding program) selected via the selection switch 459 to generate audio data VRX, and outputs it to the stereo audio generation section 453.

The stereo audio generation unit (STGEN) 453 generates audio signals @
Stereo audio signals @SR and SL are generated from V RX.

FIG. 6 is a block diagram showing the configuration of the control device 900.

Data from each input device and data to each output device are transmitted via an internal data bus 901.

That is, drawing data from the drawing input devices 100a, b, . Here, "alb..." with an alphabet added indicates that a plurality of devices with the same function are connected. (same as below).

Still image data from still image input device 200 is sent to data bus 901 via input interface 922 and data buffer 923.

Audio data from the audio input device 400 is sent to the data bus 901 via an input interface section 942 and a data buffer 943.

A frame identifier (FID) indicating which input device the data frame is from is added to these input data.

For data output, first the drawing data is transferred to the data bus 901.
from there to the flat display devices 150a, b, . . . via the output interface section 916.

In the same way, still images and drawing data are sent from the data bus 901 to the hard copy device @260 via the output interface section 926, and the output interface section 917
The video data is sent to the graphic display device (1) 160 via the output interface section 936, and the audio data is sent to the audio output device 450 via the output interface section 946. Sent.

Furthermore, still images are sent from the graphic display device (1) 160 to the data bus 901 via the input/output interface section 917.
You can also send drawing data.

Also, accumulated data from the accumulation device 600 is sent to the data bus 901 via the file interface 961.
Conversely, when storing data, the data is sent from the data bus 901 to the storage device 600 via the file interface 961.

Switching of the transmission destination of these data, selection of data, and multiplexing are performed by the frame controller 990.

When sending the data group collected by the frame controller 990 to the line, the transmission interface 972
is sent to the transmission device 700 via. A data group sent from the line via the transmission device 700 is sent to the frame controller 990 via the transmission interface 972.

Here data buffer 913.923.933.943
is used to store data in case transmission to the data bus is temporarily interrupted due to switching of data transmission destination or data selection, or when the transmission line capacity is transiently exceeded due to multiplexing. .

A conference support section 991 controls the frame controller 990. The conference support unit 991 includes a speaker identification device 510, a state detection device 800, and an operation input device 850.
The frame controller 990 selects data and selects a destination based on signals from the frame controller 990 and the like.

The amount of data input to this control device 900 is as follows.

First, the drawing data from the control devices 100a, b...
Because of the low speed of about 0.00 bps, even if multiple of these devices are installed at the same time, the data speed is about several kbps.

Therefore, the drawing data can be input at the same time, and the multiplexing unit 911 multiplexes and imports the drawing data.

Furthermore, the still image data from the still image input device 200 is 16K.
bps, and the video data from the video input device 300 is about 32 kb, it is impossible to input multiple devices at the same time when transmitting audio data to the line, and conference support is required. One of them is selected by the speaker switching signal (PID) from the section 991.

Furthermore, the audio data from the audio input device 400 is μ-LA
6 by methods such as WPCM, ADPCM, APCA8, etc.
It is used by switching from 4kpbS to 8kpbS as necessary. The switching is done by the selection signal (VRUL) from the conference support section 991.
This is supported by the voice input device 400 as follows. This signal is output by the conference support unit 991 depending on the amount of other data.

This selection signal (VRUL) is also sent to the audio output device 450, and the decoding method is determined.

Note that the input/output devices labeled with alphabets such as a, b, . . . are installed at the hands of each conference attendee.

As described above, in this embodiment, the conference support unit 991 measures the amount of traffic on the communication line, takes other data amounts into consideration, and sends a selection signal (VRUL) to the audio input device 400 and the audio output device 450. Appropriate encoding and decoding formats are selected by the device. Therefore, in this embodiment, the optimal encoding format is selected depending on the amount of traffic on the communication line.
The decoding format is selected and efficient data transmission can be performed.

[Effects of the Invention] As described in detail above, according to the present invention, the encoding format and the decoding format can be selected depending on the traffic d of the communication line, so that efficient data transmission can be performed.

[Brief explanation of drawings]

Figure 1 is a block diagram of the electronic conference system, Figure 2 is a block diagram of the audio input device, Figure 3 is a block diagram of the audio encoder, Figure 4 is a block diagram of the traffic amount of the communication line, and Figure 6. is a block diagram of a control device, and FIG. 7 is a block diagram of a conventional electronic conference system. 400...Audio input device 408...
...Audio audio encoding section 450...
Audio output device 900...Control device 991...Meeting support product Applicant Toshiba Corporation Agent Patent attorney Su Yamasa - Figure 2, Figure 4, Figure 5 Procedural amendment dated October 721, 1985 (, -', , t. Director-General's Palace 1, Indication of case Patent Application No. 159942/1981 2, Title of invention Electronic conference system 3, Person making amendment case) Patent applicant: 72 Horiyo-cho, Saiwai-ku, Yozaki City, Kanagawa Prefecture (307) Toshiba Corporation 4, Agent: 2-1 Kanda Ta-cho, Chiyoda-ku, Tokyo 101 September 30, 1988 (shipped) 6. Insert "Figure 5 is a block diagram of the audio decoding section" after "Block diagram," on page 18, line 5 of the letter of specification subject to amendment.

Claims (1)

[Claims] Drawing data, still image data, video data, audio data,
an input system that inputs the data, an output system that outputs drawing data, still image data, video data, and audio data, a transmission system that controls the transmission of these data between a remote point, and the input system and output system. , a control system for controlling a transmission system, and a plurality of conference unit systems connected via a line. An electronic conference system comprising: an audio input device that encodes in an encoding format according to the measurement result; and an audio output device that decodes audio data in a decoding format according to the measurement result and then outputs the decoded audio data. .