KR19990064542A - Method for voice communication with computer among multi user - Google Patents

Abstract

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a multi-party computer voice communication method. Conventionally, in the case of voice communication, a compression algorithm is required due to a limitation of processing and transmitting a large amount of data with a low transmission rate transmission line in a short time. Algorithm causes the load of the hardware or software of the system that performs it, resulting in the need for a server as a result of the increase in cost and the problem that the host connected to the server in the event of a failure of the server can not talk.

According to the present invention, a user clicks a call button of a counterpart to type a corresponding Internet address of the counterpart or selects and calls a counterpart's address using a phone book menu to connect a communication network between speakers, and after the communication between the speakers is connected, the micro-voice When a signal is input, the micro-inputted voice signal is encoded by a full-duplexex codec as a digital signal, and the encoded audio signal is transmitted to a high-speed Internet line through a TCP / IP packet part and built in the receiving personal computer. Transmitting to the TCP / IP packet part, selecting only the audio signal desired for simultaneous call among the plurality of audio signals transmitted to the TCP / IP packet part of the receiving personal computer, mixing using a built-in mixing function, and Restoring the original audio signal by Duplex Rex and outputting it to the speaker. It is a true invention.

Description

Multi-party computer voice communication method {METHOD FOR VOICE COMMUNICATION WITH COMPUTER AMONG MULTI USER}

The present invention relates to a multi-party computer voice communication method, and more particularly, a multi-person computer voice over a high-speed Internet in the state of removing a server for mixing a plurality of voice data or release the compressed voice data during multi-party computer voice communication The present invention relates to a multi-party computer voice communication method that enables communication.

In general, the Internet is not only domestic information but also a treasure trove of information that can be obtained easily and quickly, like information on my computer, as well as various and beneficial information spread around the world.

Today, with the development of the Internet and the rapid increase in subscribers, it is possible to transmit / receive voice over the Internet line using a microphone and a speaker mounted on a computer without using a separate telephone.

However, the sampling frequency for voice calls is usually 8Khz, and when 16bit resolution is selected using a 16bit sound card, the voice data to be transmitted becomes 128Kbit per second, which causes a heavy load on the transmission line. In addition, there was a problem in that even a slight transmission failure could easily be interrupted or other voice communication disturbances that could not be understood.

Therefore, conventionally, in order to reduce the load on the transmission line, the voice data is compressed and then transmitted, and the receiving side must go through a process of releasing the compressed voice data to restore the voice signal. Internal hardware or separate digital signal processing hardware.

In other words, the voice communication over the Internet line has a very high proportion of the voice data to be transmitted (about 8Kbps when compressed) compared to the transmission speed of the transmission line (up to 56Kbps for telephone lines and usually 256Kbps for dedicated lines). Alternatively, after compressing voice data using dedicated hardware, it has been mainly implemented in one-to-one communication.

1 is a schematic configuration diagram of a personal computer for implementing one-to-one computer voice communication according to the prior art.

Referring to FIG. 1, a ROM 10 in which a system control program is stored, a RAM 12 used for data processing, and an audio (audio) signal input from a microphone are encoded into a digital signal. 16-bit 8Khz of the coded digital signal output from the audio codec 14 and the audio codec 14 to restore the encoded digital signal transmitted to the other computer to the audio signal and output to the speaker, or transmitted from the other computer Sampling unit 16, which samples the audio signal and outputs it to audio codec 14, extracts the audio signal sampled from sampling unit 16 using a compression algorithm of about 1/10 to 1/50 of the original size. A compression and decompression unit 18 for compressing to a size or releasing a compressed audio signal transmitted from a counterpart computer, and the compressed audio output from the compression and decompression unit 18. TCP / IP (Transmission Control Protocol / Internet Protocol) packet unit 20 for outputting an audio signal to a counterpart computer via an internet line, a display unit 22 for displaying on a screen, a user interface connected to a keyboard and a mouse, and the like. And a control unit 26 for controlling the entire system.

2 is a schematic configuration diagram of an entire system for implementing multi-party computer voice communication according to the prior art.

Referring to FIG. 2, an audio signal input from a microphone is digitally encoded, sampled and compressed, and the compressed audio signal is output to the server 32 and the audio signal is output again from the server 32. It consists of a plurality of personal computer 30 to restore the output to the speaker.

The server 32 decompresses each of the voice data received from the plurality of personal computers 30, mixes the voice data of the speakers who want to simultaneously talk, and transmits the mixed data to the speakers. .

At this time, the server 32 requires dedicated hardware or software for decompression, and a function for connection or mixing between speakers is essential.

The communication method of the computer voice communication configured as described above will be described with reference to FIG.

First, in the one-to-one computer voice communication method, the user selects the Internet communication application program and the control unit 26 operates it to prepare and wait for the Internet communication transmission / reception state.

Then, the user clicks the call button to call the other party.

In addition, the user calls the correspondent by typing the corresponding Internet address from the menu displayed on the screen or using the phone book menu.

Then, the user enters a microphone voice.

At this time, the micro-input voice signal is digitally encoded by the audio codec 14.

The encoded audio signal is input to the sampling unit 16 and sampled at 16 bit 8 kHz.

The sampling frequency for the voice call is 8 kHz, and when a 16-bit resolution is selected using a 16-bit sound card, the amount of data to be transmitted per second during a voice call is as follows.

That is, the amount of audio data to be transmitted per second is 16 bits x 8000 / sec = 128 Kbps.

The audio signal sampled by the sampling unit 16 is transmitted to the compression and decompression unit 18 and compressed by a compression algorithm to a size of about 1/10 to about 1/50 of the original size.

The compressed audio signal is transmitted to the TCP / IP packet unit 20 embedded in the receiving personal computer via the Internet line.

The compressed audio signal transmitted to the TCP / IP packet part of the receiving personal computer is output to the compression and decompression unit 18. At this time, the output compressed audio signal is decompressed by dedicated hardware or software as in the case of compression. Done.

The audio signal decompressed by the compression and decompression unit 18 is restored to the original audio signal by the audio codec 14 and output to the speaker.

On the other hand, in the case where several computers simultaneously perform voice communication, in the computer voice communication method, the compressed audio signal transmitted from each computer is transmitted to the server 32.

At this time, the server 32 decompresses the received voice data and digitally mixes the voice data of the speakers who wish to make a simultaneous call and transmits the mixed audio signal to each receiving computer.

The audio signals transmitted to the receiving computers are processed in the same manner as described in the one-to-one method.

In the conventional multi-party computer voice communication method configured as described above, when the amount of voice data is not compressed for one person, 128 Kbps is assumed, and if it is assumed to be 10 simultaneous calls, 1.28 Mbps is required, and since the call is bidirectional, transmission / Considering the reception, it is 2.48Mbps, which is a very heavy load even in the case of a conventional leased line, and eventually compressing voice data is required.

However, the existing compression algorithm requires about 50% of the performance per voice channel based on IBM compatible computer Pentium 100Mhz, which puts a heavy load on the host performing compression. It is very likely that you will run into problems running applications or run compression algorithms.

In order to avoid this, it is possible to add a dedicated digital signal processor (DSP) board, but this has a problem in that the cost of adding a dedicated DSP board increases.

In addition, there is a problem that all the hosts connected with the server is in a call failure state when the server required for simultaneous multi-party call fails.

In addition, there is a problem in that additional cost of the system is increased by requiring a server for simultaneous multi-party calls.

SUMMARY OF THE INVENTION The present invention has been made to solve the above-mentioned problems of the prior art, and an object of the present invention is to use a high-speed Internet line while removing a server while creating a minimal system environment capable of removing a compression algorithm on a personal computer. The present invention provides a multi-party computer voice communication method for computer voice communication.

A feature of the multi-party computer voice communication method according to the present invention for realizing the above object is a voice communication method of a personal computer having a mixing function in a computer itself. Connecting an inter-speaker communication network by typing an internet address or selecting and calling an address of the other party using a phone book menu; when the micro voice signal is input after the inter-speaker communication network is connected, the micro-input voice signal is full-duplexed. Encoding the digital signal by a codec and transmitting the encoded audio signal through a TCP / IP packet part to a high-speed Internet line and transmitting the packet to a TCP / IP packet part embedded in a receiving personal computer. Audio wishing for simultaneous call among multiple audio signals transmitted in TCP / IP packet part Signal only selectively by mixing using a mixing function having its own and this pool - to restore the original audio signal by the Rex dyupul comprises the step of outputting to the speaker.

1 is a schematic configuration diagram of a personal computer for implementing one-to-one computer voice communication according to the prior art;

2 is a schematic configuration diagram of an entire system for implementing multi-party computer voice communication according to the prior art;

3 is a flow chart showing a multi-party computer voice communication method according to the prior art,

4 is a schematic structural diagram of an entire system for implementing multi-party computer voice communication according to the present invention;

5 is a flowchart illustrating a multi-party computer voice communication method according to the present invention.

Explanation of symbols for main parts of the drawings

50: personal computer.

Hereinafter, with reference to the accompanying drawings, preferred embodiments of a multi-party computer voice communication method according to the present invention will be described.

4 is a schematic structural diagram of an entire system for implementing multi-party computer voice communication according to the present invention.

Referring to FIG. 4, a personal computer 50 which encodes an audio signal input from a microphone provided in its own form as a digital signal and sends it to a counterpart while restoring an encoded audio signal transmitted from the counterpart and outputting the same to a speaker. ) Are each interconnected via high-speed Internet lines.

On the other hand, each of the personal computer 50 to at least build the system environment as follows to implement the present invention.

(A) PC: P-200Mhz or higher, Main Memory 16M or higher,

(B) O / S: Windows-98 (language-free),

(C) Sound card: equipped with Full-Duplex codec

(D) Lan Card: Transmission speed 10Mbps or more,

(E) Microphones and headphones: General standard.

The communication method of the multilateral computer voice communication according to the present invention configured as described above will be described with reference to FIG.

First, the user selects an Internet communication application program, and the control unit operates it to prepare and wait for an Internet communication transmission / reception state.

Then, the user clicks the call button to call the other party.

In addition, the user calls the correspondent by typing the corresponding Internet address from the menu displayed on the screen or using the phone book menu.

Then, the user enters a microphone voice.

In this case, the micro-input voice signal is encoded into a digital signal by a full-duplexx codec.

The encoded audio signal is transmitted to the TCP / IP packet part which is transmitted to the TCP / IP packet part embedded in the receiving personal computer through the high-speed Internet line.

A plurality of audio signals transmitted to the TCP / IP packet part of the receiving personal computer are selectively mixed with the audio signal of the call wisher among them, and restored to the original voice signal by the full-duplexex and output to the speaker.

The maximum number of voice communication channels through the Internet line according to the present invention is 10Mbps / (2 × 128Kbps) when using a 10Mbps transmission line is possible to implement about 39 channels.

As described above, since the present invention performs the compression algorithm by removing the compression algorithm, it is possible to remove the server by allowing the individual host to perform voice data processing or digital mixing functions that could not be performed by the individual hosts.

In addition, since the compression algorithm is removed, there is no need for dedicated software or hardware for compression and decompression, thereby reducing costs.

In addition, since there is no need for dedicated software for compression and decompression, the cause of collision with an application program due to an excessive load of the software is eliminated.

In addition, there is an effect of avoiding a call failure state between all hosts connected to the server, which occurs when a problem occurs in the server system due to the removal of the server.

Claims (1)

In the voice communication method of a personal computer having a mixing function in the computer itself, Connecting a communication network between a speaker by clicking a call button of a user and typing a corresponding Internet address of the other party or selecting and calling the other party's address using a phone book menu; When a micro voice signal is input after the inter-speaker communication network is connected, the micro input voice signal is encoded by the full-duplexx codec as a digital signal and the encoded audio signal is transmitted to the high-speed Internet line through the TCP / IP packet part. Transmitting to a TCP / IP packet part embedded in the receiving personal computer; Among the audio signals transmitted to the TCP / IP packet part of the receiving personal computer, only the audio signal desired for simultaneous call is selected and mixed by using a built-in mixing function and restored to the original voice signal by Full-Duplex. Multi-party computer voice communication method comprising the step of outputting to the speaker.