CN110035078B

CN110035078B - Audio system

Info

Publication number: CN110035078B
Application number: CN201910281539.1A
Authority: CN
Inventors: 曾世; 龙力
Original assignee: CETC 48 Research Institute
Current assignee: CETC 48 Research Institute
Priority date: 2019-04-09
Filing date: 2019-04-09
Publication date: 2021-09-21
Anticipated expiration: 2039-04-09
Also published as: CN110035078A

Abstract

The invention discloses an audio system, which comprises a client, a mobile terminal, a real-time voice server and a signaling server, wherein the client is connected with the mobile terminal through the signaling server; the client comprises a plurality of microphones; the mobile terminal is in communication connection with the real-time voice server through the signaling server; each microphone is connected with the real-time voice server and is used for carrying out real-time voice transmission so as to carry out one or more of broadcast call, group call, private call or forced call. The audio system of the invention has the advantages of simple structure, simple and convenient operation, rich functions and the like.

Description

Audio system

Technical Field

The invention mainly relates to the technical field of audio, in particular to an audio system.

Background

In the use environment of scientific research military units such as aerospace and the like, in order to ensure safety and reliability, a general conference room and a scheduling management room adopt a key-type wired microphone with audio line output, are generally powered by 12VDC or 24VDC, are provided with a chairman platform unit, can set priority and a group mode in the background, can speak for multiple people at the same time, do not have an external loudspeaker and cannot carry out point-to-point conversation.

There are two types of wired microphones for a typical lan audio system: one is audio line output and is mainly used for an independent conference room, and conference rooms at a longer distance cannot communicate with each other; the other type is an IP network microphone based on IP/TCP communication protocol function, the microphone in the form is mainly applied to broadcasting, grouping or point-to-point communication, and the defects that random grouping cannot be carried out, operation needs to be carried out in background complex software, and the operation difficulty of general workers is high.

The disadvantages of most current conference room and dispatch management room audio systems are: a call object cannot be flexibly set, background professional software needs to be logged in for setting, operation is inconvenient, and professional training is needed; the function of random switching between broadcasting and private conversation cannot be met.

Disclosure of Invention

The technical problem to be solved by the invention is as follows: aiming at the technical problems in the prior art, the invention provides the audio system which is simple in structure, simple and convenient to operate and rich in functions.

In order to solve the technical problems, the technical scheme provided by the invention is as follows:

an audio system comprises a client, a mobile terminal, a real-time voice server and a signaling server; the client comprises a plurality of microphones; the mobile terminal is in communication connection with the real-time voice server through the signaling server; each microphone is connected with the real-time voice server and is used for carrying out real-time voice transmission so as to carry out one or more of broadcast call, group call, private call or forced call.

As a further improvement of the above technical solution:

the real-time voice server comprises a microphone service component and a voice service component; the microphone service component is connected with the microphone and used for monitoring the state of the microphone; the voice service component is respectively connected with the microphone service component and the mobile terminal.

The microphone service component comprises a key processing module, a microphone event processing module, a microphone voice transceiving module and a first sound mixing module; the key processing module, the microphone event processing module and the microphone voice transceiving module are all connected with the microphone and the first sound mixing module.

The voice service component comprises an RTP receiving and transmitting module, a recording module, a user grouping management module and a second sound mixing module; the RTP receiving and transmitting module, the recording module and the user grouping management module are all connected with the second sound mixing module; the RTP receiving and transmitting module is connected with the microphone through a microphone audio push module; and the user grouping management module is connected with the mobile terminal through a TCP Client module.

The signaling server comprises a command processing module, a user management module and a database management module; the command processing module is used for analyzing the command to determine whether to log in the command or not and sending the log in command to the user management module for user verification and state maintenance; the user management module is used for inquiring or updating the data of the database management module; the database management module is used for responding to the request of the user management module.

The real-time voice server communicates with the signaling server by using a TCP (transmission control protocol); and the real-time voice server communicates with the mobile terminal by using an RTP protocol.

The system also comprises a Web server, wherein the Web server is connected with the real-time voice server.

The Web server comprises a presentation layer, an application control layer, a service logic layer, a persistence layer and a database; the presentation layer is used for displaying data and receiving data input by a user; the application control layer is used for transmitting the page request parameters to the background service logic layer, then acquiring the returned information of the service logic layer and transmitting the returned information to the page; the service logic layer is used for operating the data layer and logically processing the data service; and the persistent layer is used for being responsible for accessing the database.

The microphone is a broadcast network IP microphone with a touch screen display and is provided with a number key.

And the microphone performs voice real-time transmission with a real-time voice server through an SDK (software development kit) and a TCP (transmission control protocol).

Compared with the prior art, the invention has the advantages that:

the audio system can carry out audio conferences in different rooms in the local area network, and is flexible and convenient; the user software is simple to operate, the interface is friendly, and common workers can operate the software; the computer terminal software can be set individually according to meeting requirements, group speaking is set, users are appointed to forbid speaking, point-to-point conversation is carried out, and microphone priority is set; the conference call voice can be stored, and later-period arrangement is convenient; the system equipment adopts embedded computer processing technology and runs based on Windows database and background database, etc., and is not damaged by virus data, thus ensuring stable and reliable running of the system.

Drawings

Fig. 1 is a block diagram of the hardware configuration of the present invention.

FIG. 2 is a block diagram of the software architecture of the present invention.

Fig. 3 is a system data flow diagram of the present invention.

Fig. 4 is a block diagram of a microphone service assembly according to the present invention.

Fig. 5 is a block diagram of a signaling server according to the present invention.

FIG. 6 is a block diagram of a voice service component according to the present invention.

FIG. 7 is a flow chart of a group call in the present invention.

Fig. 8 is a flow chart of a point-to-point private call in the present invention.

Fig. 9 is a flow chart of a private call for a group call in the present invention.

Fig. 10 is a flow chart of the forced private call in the present invention.

Detailed Description

The invention is further described below with reference to the figures and the specific embodiments of the description.

As shown in fig. 1 to 10, the audio system of the present embodiment includes a client, a mobile terminal, a real-time voice server, and a signaling server; the client comprises a plurality of microphones; the mobile terminal is in communication connection with the real-time voice server through the signaling server; each microphone is connected with the real-time voice server and is used for carrying out real-time voice transmission so as to carry out one or more of broadcast call, group call, private call or forced call. Specifically, the hardware part of the audio system mainly comprises a switch, a server, a network media matrix, a sound box, a sound console, a microphone, a computer and the like, and the software part mainly comprises a voice consultation system written in JAVA language. The IP network broadcast is a network two-way talkback voice communication system which combines the network communication technology, adopts the digital IP/TCP communication protocol based on the network transmission and transmits the audio signals in the form of data packets on the local area network and the wide area network based on the network digital audio technology, and combines an audio server and operation hardware to form a pure digital transmission network. The selected wired microphone is a broadcast type network IP wired microphone LS-9702A with a 7-inch touch screen display, and the device is provided with a number key and a function key interface; the audio signal with 16-bit CD tone quality is transmitted in a network mode by the aid of a high-fidelity loudspeaker, supporting TCP/IP, UDP and IGMP (multicast group control protocol). The power amplifier, the network media matrix, the sound console, the audio server and the core switch are placed in the audio cabinet of the conference room for integration. The access switches of all the call rooms are connected with the core switch in the conference room through optical fibers, so that the whole system is in a local area network, and a powerful digital communication system can be realized through software only by accessing a terminal computer into the local area network.

As shown in fig. 2, the main functions and architectures of the software are: through the software of design, the function that can directly realize on wired microphone and computer terminal has:

(1) creating a group and managing the microphone devices in the group;

(2) the broadcasting function is that a user can specify a microphone to speak and broadcast through a power amplifier sound box;

(3) the point-to-point communication function enables a user to dial through a wired microphone or computer software and communicate with any single or multiple wired microphone ends.

(4) The administrator can directly adjust the volume of any sound box and any microphone in the background;

in this embodiment, the web server may be divided into 4 layers from the foreground to the background:

1) and (6) a presentation layer. Namely, the interface layer mainly functions to display data and receive data input by a user, and provides an interactive operation interface for the user, and the main technology is ExtJS.

2) A control layer is applied. The main function is to transmit the page request parameters to the background service layer, then obtain the service layer return information and transmit it to the page, and the use of Struts2 framework is used to realize.

3) And a service logic layer. Mainly aiming at the operation of specific business problems, the method can also be understood as the operation on a data layer, and the data business logic is processed, if the data layer is a building block, the logic layer builds the building block, and the method is realized by using a Spring framework.

4) And a durable layer. Namely a data access layer, the main function of which is responsible for accessing the database and is realized by using a Druid Ali database connection pool.

As shown in fig. 3, in the present embodiment, the real-time communication system is a C/S architecture and is divided into two parts, namely a client and a server. The client is mainly an IP paging microphone and other devices. The server includes: a signaling server and a real-time voice server.

1) An IP paging microphone. And carrying out voice real-time transmission with a real-time voice server by using the SDK and the TCP.

2) A signaling server. And the signaling transmission control responsible for real-time communication uses a TCP protocol. The underlying TCP communication is realized mainly by using a libuv library, Google is used as a log system, and MySQL Connector/C + + is used for accessing the Mysql database.

3) A real-time voice server. The system is responsible for real-time voice data transmission, uses a websocket protocol to communicate with a web page, uses JSON for the data transmission format of the websocket, uses RapidJSON for the parsing library, uses a TCP protocol to communicate with a signaling server, and uses an RTP protocol to communicate with a mobile terminal. The underlying TCP communication is realized mainly by using a libuv library, Google is used as a log system, and MySQL Connector/C + + is used for accessing the Mysql database. RTP transport was implemented using a jtplib library, Speex: jitter buffering of the Jitter Buffer of the real-time data of the Jitter Buffer uses Crypto + + real-time voice data encryption and decryption.

In this embodiment, the corresponding call functions are:

(1) the group call flow, as shown in fig. 7; after a user logs in and clicks to enter a group, a broadcast key is pressed, and then group speaking can be carried out; lifting the broadcast button to end the group call;

(2) the user answers the call flow: when the call comes in, the user clicks and accepts, and enters point-to-point private conversation, and the specific flow is shown in fig. 8;

(3) group call, selecting private call of member in group, the specific flow is shown in fig. 9; dialing one member of the group, and starting conversation if the member is connected; if not, the mobile terminal is still in a group call state;

(4) dialing and forced calling flow: selecting a user to carry out private conversation, and carrying out the private conversation if the called user accepts; if the called user rejects, but the priority of the current calling user is higher than that of the called user, the private conversation can be carried out by forced calling. The specific flow is shown in fig. 10;

in this embodiment, a data flow diagram of a server system is shown in fig. 3, and the system is divided into a real-time voice server, a signaling server, a web server (the web system in fig. 3), and a paging microphone;

1) a microphone service component, as shown in FIG. 4; the microphone service component comprises a key processing module, a microphone event processing module, a microphone voice transceiving module and a first sound mixing module; the key processing module, the microphone event processing module and the microphone voice transceiving module are all connected with the microphone and the first sound mixing module;

the keys reported by the microphone can be received through the microphone SDK and are analyzed and processed by the key processing module;

the state events of the microphone, including switching events of states such as connection, disconnection, conversation, idle and the like, can be monitored through event callback of the microphone SDK, and are delivered to a microphone event processing module to execute corresponding operations;

the voice data is recalled back through the microphone SDK and transmitted to the microphone voice receiving and transmitting module, the received voice data is transmitted to the first voice mixing module to be mixed, and then the voice data returns to the microphone voice receiving and transmitting module and is transmitted back to the corresponding microphone.

2) A signaling service component, as shown in FIG. 5; the signaling server comprises a command processing module, a user management module and a database management module; the signaling server has the main functions of login authentication of a user and user command forwarding; the external command is sent to the command processing module through a TCP protocol, corresponding processing is carried out after the command is analyzed, if the command is a login command, the user management module is switched to, and user verification and state maintenance are carried out by the user management module. If the command is other commands, forwarding the command to the real-time voice server;

3) a voice services component, as shown in FIG. 10; the voice service component comprises an RTP receiving and transmitting module, a recording module, a user grouping management module and a second sound mixing module; the RTP receiving and transmitting module, the recording module and the user grouping management module are all connected with the second sound mixing module; the RTP receiving and transmitting module is connected with the microphone through the microphone audio push module; and the user grouping management module is connected with the mobile terminal through the TCP Client module.

Although the present invention has been described with reference to the preferred embodiments, it is not intended to be limited thereto. Those skilled in the art can make numerous possible variations and modifications to the present invention, or modify equivalent embodiments to equivalent variations, without departing from the scope of the invention, using the teachings disclosed above. Therefore, any simple modification, equivalent change and modification made to the above embodiments according to the technical spirit of the present invention should fall within the protection scope of the technical scheme of the present invention, unless the technical spirit of the present invention departs from the content of the technical scheme of the present invention.

Claims

1. An audio system is characterized by comprising a client, a mobile terminal, a real-time voice server and a signaling server; the client comprises a plurality of microphones; the mobile terminal is in communication connection with the real-time voice server through the signaling server; each microphone is connected with the real-time voice server and is used for carrying out real-time voice transmission so as to carry out one or more of broadcast call, group call, private call and forced call;

the real-time voice server comprises a microphone service component and a voice service component; the microphone service component is connected with the microphone and used for monitoring the state of the microphone; the voice service component is respectively connected with the microphone service component and the mobile terminal;

the microphone service component comprises a key processing module, a microphone event processing module, a microphone voice transceiving module and a first sound mixing module; the key processing module, the microphone event processing module and the microphone voice transceiving module are all connected with the microphone and the first sound mixing module;

2. The audio system according to claim 1, wherein the signaling server comprises a command processing module, a user management module, and a database management module; the command processing module is used for analyzing the command to determine whether to log in the command or not and sending the log in command to the user management module for user verification and state maintenance; the user management module is used for inquiring or updating the data of the database management module; the database management module is used for responding to the request of the user management module.

3. The audio system according to any one of claims 1 to 2, wherein the real-time voice server communicates with the signaling server using a TCP protocol; and the real-time voice server communicates with the mobile terminal by using an RTP protocol.

4. The audio system according to any one of claims 1 to 2, further comprising a Web server, said Web server being connected to said real-time voice server.

5. The audio system of claim 4, wherein the Web server comprises a presentation layer, an application control layer, a business logic layer, a persistence layer, and a database; the presentation layer is used for displaying data and receiving data input by a user; the application control layer is used for transmitting the page request parameters to the background service logic layer, then acquiring the returned information of the service logic layer and transmitting the returned information to the page; the business logic layer is used for operating the database and logically processing the data business; and the persistent layer is used for being responsible for accessing the database.

6. The audio system according to any one of claims 1 to 2, wherein the microphone is a broadcast network IP microphone with a touch screen display and is provided with a number key.

7. The audio system according to any one of claims 1 to 2, wherein the microphone performs voice real-time transmission with a real-time voice server via SDK and TCP protocols.