KR100343491B1 - Sound Transmission System for real time and ambidirection in using sound - Google Patents

Abstract

The present invention relates to a system for reproducing without a sound collision while using a computer sound card. In a PC-to-PC Internet call system connected to a network, a request is made from a caller's system to a caller's system, and is accepted and connected to each other. First step (S1); Preparing a buffer for recording a certain amount of voice in the talker's system (S2); A third step (S3) of preparing a buffer for reproducing a predetermined amount of voice in the system of the called party; A fourth step (S4) in which the caller inputs a voice through a microphone and stores the voice in a voice recording buffer in a caller's system and transmits the call to the caller's system using UDP; A fifth step (S5) of receiving the data transmitted in the fourth step to the voice reproducing buffer in the system of the called party and reproducing the voice; A sixth step (S6) in which, after the initial voice call is started, as many users as the number of users having a buffer for voice recording and a buffer for voice playback are connected to the system of the caller and the receiver; A seventh step (S7) for placing the recorded and reproduced buffer at the end of the queue so that it can be used next time in the system of the talker and the receiver. It is about. And, the first step (S1) is characterized in that the direct connection without passing through the server in a manner that is connected by inputting the IP address of the caller, the called party. In addition, the fifth step (S5) is characterized in that the sound recording buffer transmitted in the fourth step (S4) by collecting a predetermined buffer or more to ensure that there is no interruption of the sound. In the seventh step S7, the received voice recording buffer is reproduced as real-time voice data and placed in an empty buffer in the last queue, thereby preventing the voice data from accumulating in the buffer. Even when using the sound is characterized in that the call can be made without interruption of the voice.

Description

Sound Transmission System for real time and ambidirection in using sound}

The present invention relates to a system that plays without sound collision while using a computer sound card. In a PC-to-PC Internet call system connected to a network, the present invention is suitable for a multi-party call and a one-to-one call without conflict with other sounds such as game sounds. It relates to a system that can make calls.

In general, Internet phones are classified into a PC-to-Phone method, a Phone-to-Phone method, and a PC-to-PC method.

First, in PC-to-Phone method, visit the website of the ISP company, download the program, install the program on the user's computer, enter the corresponding phone number to call and press the call button. Connected to a mobile phone, etc., you can talk. When connected in the above manner, the user talks through a microphone, which is a peripheral device of the computer, and receives a sign through the speaker. In addition, the above-described method is connected through an ISP company, and the ISP company operates in a manner of collecting a fee from a user, so that the user must pay for the telephone while using the Internet line. In addition, the PC-to-Phone method has a problem that no other sound can be used during a telephone call.

Next, the Phone-to-Phone method is a method that has been used in recent years as a method of using the Internet phone using the phone. In the above method, the fee for use is lower than that of a general international call, but there is a disadvantage in that a general telephone cannot be used, and the telephone itself must be provided separately and the telephone itself is expensive.

Next, the PC-to-PC method is the same as the above two methods in the method using the Internet line, but is a mutual call method using a PC. The above method is most famous for the company product called VocalTec by using a PC's microphone, sound card, speaker, and corresponding software. In the case of Bocaltech, the PC-to-PC method is widely known to users as a method of connecting to a server computer of Bocaltech and the server connection success rate is decreasing. Therefore, there is a problem that it is difficult to make a direct call at the desired time. In addition, since the server utilization is high, there is also a problem of poor communication due to the slow speed. In addition, in the case of the Vocaltec Internet phone, there is another problem that other sounds cannot be used during a call.

In addition, there is a voice message among systems for delivering voice between computers. The voice message is the same as an e-mail that is commonly used by general Internet users. The voice message is stored by the caller as a voice and sent to the corresponding e-mail address. The above method is not a real-time transmission method, but merely a conversion of an e-mail, which is a text message method, to a voice message method.

In addition, in the case of a game room that has become popular in recent years, games are often carried out in groups. In proceeding with such a game, in most cases, the intention is to communicate each other by voice pronunciation. As described above, the phonetic pronunciation not only causes significant discomfort to others playing around,

In order to solve the above problems, a mutual call program is used. However, the call program may collide with an effect sound generated during a game, and thus voice may not be properly transmitted. Therefore, a person who receives a voice often does not understand the content to be delivered.

The present invention for solving the above problems is to provide a voice transmission system that does not affect other sounds in the system of the PC-to-PC call between the separation distance.

It is also an object of the present invention to provide a system that enables direct one-to-one or many-to-many calls without going through a server computer, and provides a system in which voice data does not collide with each other. Therefore, the talker and the talker can connect to each other at a desired time and can talk to each other without a sound collision during the game.

1 is an overall configuration diagram of a real-time, two-way speech transmission system during sound use according to the present invention.

2 is a structure diagram of a stream buffer allocation structure in the form of a win type queue for recording voice data of a real-time, two-way voice transmission system during sound use according to the present invention.

3 is a circular queue buffer operation and voice data transmission diagram for recording voice data of a real-time, two-way voice transmission system during sound use according to the present invention.

4 is a flowchart for generating a direct sound buffer for recording voice data of a real-time, two-way voice transmission system and recording the received voice in a direct sound buffer during sound use according to the present invention.

5 is a flowchart of recording and reproducing voice data into a stream buffer of direct sound of a real-time, two-way voice communication system during sound use according to the present invention.

In order to achieve the above object, the present invention provides a PC-to-PC Internet call system connected to a network, comprising: a first step of requesting a connection from a caller's system to a caller's system, and receiving and accepting a connection; Preparing a buffer for recording a certain amount of voice in the talker's system; Preparing a buffer for reproducing a certain amount of voice in the system of the called party; A fourth step of the caller inputting a voice through a microphone, and storing the voice in a voice recording buffer in a caller's system and transmitting the call to the caller's system using UDP; A fifth step of receiving the data transmitted in the fourth step into the voice reproducing buffer in the system of the called party and reproducing the voice; A sixth step after the initial voice call is started, each number of users having a buffer for voice recording and a buffer for voice playback being connected to the system of the caller and the receiver; And a seventh step of placing the recorded and reproduced buffer at the end of the queue so that it can be used later in the system of the talker and the receiver.

And, the first step is characterized in that the direct connection without passing through the server in a manner that is connected by inputting the IP address of the caller, the called party.

In addition, the fifth step is characterized in that the sound recording buffer transmitted in the fourth step is collected by playing a predetermined buffer or more so that there is no interruption of the sound.

In the seventh step, the received voice recording buffer is reproduced as real-time voice data and placed in the last queue as an empty buffer, thereby preventing the voice data from accumulating in the buffer. It is characterized by being able to talk without interruption of voice even when using.

Accordingly, the user can connect 1: 1 or many-to-many communication directly to the other party's computer without making a sound collision even during a game without connecting to the server computer.

Hereinafter, with reference to the accompanying drawings, it will be described in detail the real-time, two-way voice transmission system in use of the sound according to the present invention.

1 is an overall configuration diagram of a real-time, two-way voice communication system during sound use according to the present invention, and FIG. to be. 3 is a circular queue buffer operation and voice data transmission diagram for recording voice data of a real-time, two-way voice transmission system according to the present invention, and FIG. FIG. 5 is a flowchart for generating a direct sound buffer for recording voice data and recording the received voice in the direct sound buffer. The playback flow chart.

First, the first step S1 will be described. In the PC-to-PC Internet call system, a connection request is made from a caller's system to a caller's system, and is accepted. In more detail, the sender's system first sends a connection request signal, and the receiver's system receiving the connection request signal sends a connection valid message. Accordingly, the sender's system sends a connection acceptance message and is ready to talk next. In addition, the connection method in the first step is fundamentally different from the server connection by the IP connection to the counterpart system. In other words, they are connected in the concept of 1: 1, not server-client.

Next, a second step (S2) will be described. The second step is to prepare a buffer for recording a certain amount of voice in the talker's system in order to transmit the talker's voice while the caller and the talker system are connected to each other in the first step. If there is only one receiver, the buffer will be prepared for about one second. The 1 second buffer means 10 buffers of a queue type (hereinafter referred to as a queue) of 0.1 seconds.

Next, a third step S3 will be described. In the same manner as in the second step, the receiver's system prepares a buffer for a certain amount of speech reproduction in the receiver's system in order to receive the speaker's voice. As in the second step, when there is only one talker, the buffer is prepared for about one second. The 1 second buffer means 10 buffers of a queue type (hereinafter referred to as a queue) of 0.1 seconds.

Next, a fourth step S4 will be described. The fourth step is a step in which the talker inputs a voice through a microphone, and the sender's system stores the input voice in a voice recording buffer and transmits it to the talker's system using UDP (User Data Protocol). The UDP transmission method is faster than a general TCP transmission method. The reason is that the UDP transmission method is a structure that transmits data without connection setting, error control, and flow control unlike TCP transmission.

Next, a fifth step S5 will be described. Receiving the data transmitted in the fourth step to the voice reproduction buffer in the system of the called party to reproduce the voice. The voice reproduction is received by the voice reproduction buffer of the talker system and passed to the direct sound to reproduce the voice. In addition, by reproducing the voice recording buffer transmitted in the fourth step by collecting a predetermined buffer or more, there is no interruption of the sound. For example, if a cue of 0.1 seconds is transmitted from the talker's system, the caller's system can prevent the voice breakup when the cues are collected and reproduced in units of 3 to 4 (0.3 to 0.4 seconds). .

Next, a sixth step S6 will be described. In the sixth step, after the first voice call is started, the number of users who have a buffer for voice recording and a buffer for voice playback are connected to the system of the caller and the receiver. In this step, since the call between the talker and the talker must be made, the talker's system creates a buffer for recording the voice and a voice playback for the number of the target parties. In this manner, the system of the called party creates a buffer for recording voices and a buffer for playing voices as many as the number of target parties.

Next, a seventh step S7 will be described. The seventh step is to place the recorded and reproduced buffer at the end of the queue in the system of the talker and the receiver.

Due to the above steps it is possible to use without a conflict even when using other sounds.

Hereinafter, an embodiment using a real-time, two-way voice communication system while using the sound according to the present invention will be described.

First, the user installs the software provided with the real-time, two-way voice communication system while using the sound according to the present invention. Next, enter the IP address of the target to call. In this case, the user may have one object or multiple objects. And, while the user waits or before inputting the IP address, the user already runs the game program and enjoys the game while watching the sound of the game program.

Next, when the user presses the connection button, the system according to the present invention finds the corresponding IP address. The IP address finds the network through the Address Resolution Protocol (ARP). Then, it finds the network and sends a connection request message. The called party system receiving the connection request message sends the connection valid message back to the sender's system, and the system of the called party receiving the connected valid message transmits the connection acceptance message. From this point on, the talker can talk to the called party. The voice data first spoken by the talker is transmitted in a packet format in the talker system, and the delivered message is recorded using a queue-type stream buffer in the talker's system. The system uses direct sound to mix and play. Therefore, the speaker or earphone of the receiver system is transmitted to the receiver's ear.

In the seventh step, simultaneous calls between the talker and the receiver are sent to the last position of the buffer where the recorded and reproduced cues are sent. By doing the above, the queue is configured in the form of a circular queue. Done. With the above configuration, the contents of the queue can be reproduced or recorded without immediately accumulating the contents of the queue, and the contents can be deleted immediately, thereby reproducing and recording the voice without sound collision.

As described above, according to the present invention, according to the real-time, two-way voice delivery system, the continuous use of the queue on the buffer in the form of a circular queue, so that the voice is delivered without sound collision, so that there is no inconvenience in the game. The purpose is to provide a system that can

Claims (4)

In the PC-to-PC Internet call system connected to the network, Requesting a connection from the talker's system to the talker's system, and receiving and accepting the first step (S1); Preparing a buffer for recording a certain amount of voice in the talker's system (S2); A third step S3 of preparing a buffer for reproducing a certain amount of voice in the system of the called party; A fourth step (S4) in which the caller inputs a voice through a microphone and stores the voice in a voice recording buffer in a caller's system and transmits the call to the caller's system using UDP; A fifth step (S5) of receiving the data transmitted in the fourth step to the voice reproducing buffer in the system of the called party and reproducing the voice; A sixth step (S6) in which, after the initial voice call is started, as many users as the number of users having a buffer for voice recording and a buffer for voice playback are connected to the system of the caller and the receiver; A seventh step (S7) for placing the recorded and reproduced buffer at the end of the queue so that it can be used next time in the system of the talker and the receiver. . The system of claim 1, wherein the first step is directly connected to the caller and the receiver by inputting an IP address and not through a server. [5] The system of claim 1, wherein the fifth step reproduces the voice recording buffer transmitted in the fourth step by collecting a predetermined buffer or more so that there is no interruption of sound. The method of claim 1, wherein the seventh step is performed by reproducing the received voice recording buffer as real-time voice data and placing the empty buffer in the last queue, thereby preventing the voice data from accumulating in the buffer. Real-time, two-way voice delivery system using the sound, characterized in that the voice can be called without interruption even when using the sound of the game.