Audio call format applying apparatus and method of naval integrated communication system

Abstract

The present invention relates to an apparatus and method for applying a voice call format in an Integrated Communication System (ICS) for a vessel, implemented to improve voice quality by processing high-quality voice signals while minimizing call delay by applying a high-quality voice call format in an ICS environment for a vessel. The apparatus comprises: an audio mixer controller performing audio mixing settings to control the transmission and reception of a format-applied audio signal and transmitting the configured audio mixing information; an audio mixer receiving a format-applied audio signal according to the audio mixing information transmitted from the audio mixer controller, mixing the signal, and transmitting the mixed format-applied audio signal; a communication terminal applying a voice call format to an analog audio signal received through an input means, converting it into a format-applied audio signal, and transmitting it to the audio mixer, and receiving a format-applied audio signal transmitted from the audio mixer, converting it into an analog audio signal, and outputting it through an output means; and an analog interlocking terminal applying a voice call format to an analog audio signal, receiving the analog audio signal, converting it into a format-applied audio signal, and transmitting it to the audio mixer, and receiving a format-applied audio signal transmitted from the audio mixer, converting it into an analog audio signal, and transmitting it. An analog sound device receives an analog audio signal through an input means and transmits it to an analog linked terminal, and outputs the analog audio signal transmitted from the analog linked terminal through an output means.

Description

Technology Field

[0001] The technical field of the present invention relates to an apparatus and method for applying a voice call format in an integrated communication system for ships, and in particular, to an apparatus and method for applying a voice call format in an integrated communication system for ships that is implemented to improve voice quality by processing high-quality voice signals while minimizing delay during a call by applying a high-quality voice call format in an Integrated Communication System (ICS) environment for ships. Background Technology

[0002] The Integrated Communication System (ICS) for ships is a system that operates a general integrated communication system on a ship. It operates a complex voice communication system to transmit operational situations and tactical information to necessary locations within the ship for the purpose of command and control, and to transmit collected tactical information and operational details to the outside of the ship when necessary. This Integrated Communication System for ships is operated by integrating it into user terminals of the integrated communication system (i.e., intercom terminals). As a Non-IP based system, it consists of an in-ship communication system, an out-of-ship communication system, a lunar telephone system, a broadcasting and alarm system, an in-ship wireless communication system, and a lunar telephone system. It is primarily operated for voice communication, and various configurations are possible depending on the mission and characteristics of the ship.

[0003] The external communication system consists of various types of communicators for each frequency band, a digital message handling system (DMHS) for message processing, security equipment for encryption processing, and modems. The operation method of the communicators for each frequency band can be changed at any time according to the military's objectives and requirements, and serves the role of wirelessly transmitting and receiving operational, command, and tactical information from land bases, ships, and aircraft across various frequency bands. Furthermore, to prevent eavesdropping, all external communications are operated to be transmitted and received via security equipment during operations, and are also linked with digital recording devices to allow for the recording of communication content when necessary for the analysis of communication details and operational situations.

[0004] The broadcasting and alarm system consists of a broadcasting control device, a remote operating device, an amplifier, and speakers for generating alarm sounds, monitoring the status of remote operating devices, and selecting entertainment broadcasts (e.g., CDs, music, or general radio broadcasts), and is a system that enables broadcasting, alarms, and entertainment broadcasts to key areas of the vessel through speakers for each zone or anti-ship speakers.

[0005] The onboard wireless communication system consists of wireless relay devices, wireless terminals, radial coaxial cables, and antennas, and is used for disaster control, maritime resupply, helicopter control, engine room operations, and maintenance control. In particular, to overcome the poor wireless environment inside the vessel, radial coaxial cables are laid in key operational areas and antennas are placed at key locations to operate wireless terminals.

[0006] The lunar telephone system is a communication network designed to operate using built-in batteries or lunar power without the need for a separate power supply to the equipment. It consists of battery-powered lunar telephone headsets, distributors, and headset connection sockets, and is distributed throughout key areas of the vessel for missions such as vessel control, entry and exit from port and mooring, and resupply operations.

[0007] Korean Registered Patent No. 10-2733289 (registered on November 19, 2024) discloses an IP-based integrated communication system for ships and a communication method for the integrated communication system for ships. It is characterized by comprising an integrated interlocking device for interlocking Non-IP equipment, which is an integrated component of an existing integrated communication system, and an IP interlocking gateway, which is a new integrated component, based on the configuration of the integrated interlocking device, an IP-based integrated communication system including an IP-based switch, and a ship integrated network communication unit. The ship integrated network communication unit additionally includes a communication network management function for monitoring the operational status of the integrated communication network and user terminals, an S-ICS server for installing a software security solution, and an S-ICS gateway for information protection between various communication systems with different security grades. The S-ICS gateway of the ship integrated network communication unit is configured to use one or more of a cross-domain solution (CDS), a software-defined perimeter (SDP), and a software-defined network (SDN) on the internal and external boundary networks centered on the integrated communication system.According to the disclosed technology, the system is designed to enable interoperability by linking various multimedia-oriented operating systems, including existing non-IP-based voice signal-oriented communication systems and IP-based data and video information operating on the ship. It is designed to be linked with the combat system by integrating an IP-based integrated communication system and an information protection system so that the transmission and blocking of information with different security levels collected by each communication system can be achieved. Furthermore, a backbone-centered integrated ship communication network is designed by classifying the communication lines of the ship's various operating systems and communication systems by security level. In the case of communication systems, a communication network integrating voice and data is designed centered on the integrated communication system, thereby providing superior security and interoperability. Additionally, by designing an integrated ship network that integrates the ship's network, the same network line can be used in all compartments within the ship, thereby improving installation, maintenance, and operability. This enhances the convenience of the communication system to make it more suitable for executing operational command decisions, thereby enabling effective operational execution.

[0008] Korean Published Patent No. 10-2015-0056120 (published May 26, 2015) discloses an audio output control method and an apparatus supporting the same, which can provide three-dimensional sound services by generating an audio signal modified to match the channel characteristics of a multi-channel audio source file to the characteristics of a plurality of audio output devices and playing the audio signal assigned to each of the plurality of audio output devices, can provide sound services optimized according to user requirements by extracting an audio signal from a sound source and generating a modified audio signal, and can provide sound services modified in various forms according to the positional configuration of the plurality of audio output devices. According to the disclosed technology, in an audio output control method for an audio output device that is communicationally connected to at least one other audio output device, the method comprises: a process of converting specific audio data to generate at least one modified audio signal assigned to the audio output device and at least one other audio output device; and a process of playing at least one of the generated modified audio signals to a channel assigned to the audio output device. The method includes the process of transmitting the remainder of the generated modified audio signal to at least one other audio output device, wherein the modified audio signal is a signal in which at least one of the sound source and channel characteristics of specific audio data is altered.

[0009] In the conventional technology described above, telephone systems perform one-on-one calls between each terminal, group calls among multiple terminals, receive the same voice in mono or stereo, and proceed with the call by entering a call number. In the case of VoIP, it is an IP-based telephone system that performs telephone calls based on Ethernet, allows for a voice delay of less than 300ms round trip, and digitizes and transmits voice through 8,000 voice samples per second. Additionally, in the case of intercoms, calls and conversations are performed based on buttons, and have the disadvantage that the connection assignments of the buttons are fixed and cannot be changed through program settings.

[0010] In the conventional technology described above, when converting analog audio sources into digital audio sources during communication on naval vessels, 8,000 samples are collected per second; that is, by sampling analog voice 8,000 times per second, there is a disadvantage of a low maximum frequency range that can be processed. Additionally, since the expressible range is 8 bits—meaning data is processed within an 8-bit range—there are only 256 values, resulting in a disadvantage of low precision. Furthermore, since mixing is performed after converting to a 32-bit Float type, there is a delay during voice mixing, and there are disadvantages that make it difficult to use SONAR and play high-quality audio sources. Prior art literature

[0011] Korean Registered Patent No. 10-2733289, Korean Published Patent No. 10-2015-0056120 The problem to be solved

[0012] The problem that the present invention aims to solve is to overcome the aforementioned disadvantages by providing an apparatus and method for applying a voice call format in an Integrated Communication System (ICS) for ships, which is implemented to improve voice quality by processing high-quality voice signals while minimizing delay during a call by applying a high-quality voice call format in an Integrated Communication System (ICS) environment for ships. means of solving the problem

[0013] As a means for solving the above-mentioned problem, according to one feature of the present invention, there is provided a voice call format application device in an integrated communication system for a vessel, comprising: an audio mixer controller that performs audio mixing settings for controlling the transmission and reception of a format-applied audio signal and transmits the set audio mixing information; an audio mixer that receives a format-applied audio signal according to the audio mixing information transmitted from the audio mixer controller, mixes it, and transmits the mixed format-applied audio signal; a communication terminal that applies a voice call format to an analog audio signal received through an input means, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives a format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and outputs it through an output means; an analog interlocking terminal that receives an analog audio signal applied to a voice call format, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives a format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and transmits it; and an analog sound device that receives an analog audio signal through an input means, transmits it to the analog interlocking terminal, and outputs an analog audio signal transmitted from the analog interlocking terminal through an output means.

[0014] In one embodiment, the audio mixer is characterized by mixing immediately without data conversion during the mixing operation using 44,100Hz sampling and 32-bit float data received from the communication terminal or the analog interlocking terminal.

[0015] In one embodiment, the communication terminal is characterized by applying a floating-point based 32-bit PCM voice format in packet-based communication as the voice call format.

[0016] In one embodiment, the communication terminal is characterized by applying the floating-point-based 32-bit PCM voice format during analog audio signal processing to convert it into 44,100Hz sampling and 32-bit float data.

[0017] In one embodiment, the communication terminal is characterized by sampling 48,000 times per second during analog audio signal processing, generating float data in the form of 32 bits, and transmitting it to the audio mixer.

[0018] In one embodiment, the communication terminal is characterized by being coupled with a packet-based real-time communication structure in the case of a floating-point-based 32-bit PCM voice format.

[0019] In one embodiment, the communication terminal is characterized by transmitting in conjunction with the voice packet processing flow of the ICS in the case of a floating-point based 32-bit PCM voice format.

[0020] In one embodiment, the communication terminal is characterized by encoding, decoding, and streaming in real time in a high-speed network environment in the case of a floating-point-based 32-bit PCM voice format.

[0021] In one embodiment, the communication terminal is characterized by applying a delay correction algorithm to account for packet loss in the case of a floating-point-based 32-bit PCM voice format and transmitting it.

[0022] In one embodiment, the communication terminal is characterized by minimizing and applying format conversion and compression steps during voice format processing by utilizing lightweight processing logic for delay minimization.

[0023] In one embodiment, the analog interlocking terminal is characterized by applying a floating-point based 32-bit PCM voice format in packet-based communication as the voice call format.

[0024] In one embodiment, the analog interlocking terminal is characterized by applying the floating-point-based 32-bit PCM voice format during analog audio signal processing to convert it into 44,100Hz sampling and 32-bit float data.

[0025] In one embodiment, the analog interlocking terminal is characterized by sampling 48,000 times per second during analog audio signal processing, generating float data in a 32-bit format, and transmitting it to the audio mixer.

[0026] In one embodiment, the analog interlocking terminal is characterized by being linked to a packet-based real-time communication structure in the case of a floating-point-based 32-bit PCM voice format.

[0027] In one embodiment, the analog interlocking terminal is characterized by transmitting in conjunction with the voice packet processing flow of the ICS in the case of a floating-point based 32-bit PCM voice format.

[0028] In one embodiment, the analog interlocking terminal is characterized by encoding, decoding, and streaming in real time in a high-speed network environment in the case of a floating-point-based 32-bit PCM voice format.

[0029] In one embodiment, the analog interlocking terminal is characterized by applying a delay correction algorithm to account for packet loss in the case of a floating-point-based 32-bit PCM voice format and transmitting it.

[0030] In one embodiment, the analog interlocking terminal is characterized by minimizing and applying format conversion and compression steps during voice format processing by utilizing lightweight processing logic for delay minimization.

[0031] In one embodiment, the voice call format application device in the integrated communication system for the vessel further comprises: a network switch formed between the audio mixer and the communication terminal to switch a format-applied sonar signal or music data between the audio mixer and the communication terminal; and a sonar device that receives an analog sonar signal through an input means and transmits it to the analog linked terminal, and outputs an analog sonar signal transmitted from the analog linked terminal through an output means.

[0032] In one embodiment, the voice call format application device in the integrated communication system for the vessel further comprises a music device that receives analog music data through an input means and transmits it to the analog linked terminal, and outputs the analog music data transmitted from the analog linked terminal through an output means.

[0033] In one embodiment, the audio mixer controller is characterized by performing sonar or music mixing settings to control the transmission and reception of a format-applied sonar signal or music data, and transmitting the set sonar or music mixing information to the audio mixer.

[0034] In one embodiment, the audio mixer receives a format-applied sonar signal or music data transmitted from the network switch or the analog interconnected terminal according to sonar or music mixing information transmitted from the audio mixer controller, mixes the data, and transmits the mixed format-applied sonar signal or music data to the analog interconnected terminal or the network switch.

[0035] In one embodiment, the audio mixer exchanges voice with a communication terminal, and the audio mixer is characterized by mixing directly without data conversion during the mixing operation using 44,100Hz sampling and 32-bit float data received from the network switch or the analog interconnected terminal that passes through it in the middle.

[0036] In one embodiment, the communication terminal is characterized by monitoring the analog sonar signal of the sonar device or the music data of the music device from a remote location at a distance.

[0037] In one embodiment, the communication terminal is characterized by applying a floating-point-based 32-bit PCM voice format when processing analog sonar signals or music data, and converting it into 44,100Hz sampling and 32-bit float data.

[0038] In one embodiment, the analog interlocking terminal is characterized by applying a floating-point-based 32-bit PCM voice format when processing analog sonar signals or music data, and converting it into 44,100Hz sampling and 32-bit float data.

[0039] As a means for solving the above-mentioned problem, according to another feature of the present invention, the method comprises: a step in which an audio mixer controller performs audio mixing settings to control the transmission and reception of a format-applied audio signal and transmits the configured audio mixing information; a step in which an audio mixer receives a format-applied audio signal according to the audio mixing information transmitted from the audio mixer controller, mixes it, and transmits the mixed format-applied audio signal; a step in which a communication terminal applies a voice call format to an analog audio signal received through an input means, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives the format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and outputs it through an output means; a step in which an analog-linked terminal applies a voice call format to an analog audio signal, receives it, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives the format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and transmits it. The present invention provides a method for applying a voice call format in an integrated communication system for a vessel, comprising the step of an analog sound device receiving an analog audio signal through an input means and transmitting it to the analog linked terminal, and outputting an analog audio signal transmitted from the analog linked terminal through an output means. Effects of the invention

[0040] The effect of the present invention is that by providing a device and method for applying a voice call format in an Integrated Communication System (ICS) for ships, which is implemented to improve voice quality by processing high-quality voice signals while minimizing delay during a call by applying a high-quality voice call format in an ICS environment for ships, it is possible to achieve improved voice resolution and minimize distortion compared to existing methods by applying Float 32-bit 48,000 Sample PCM, enable the implementation of low-latency calls capable of transmitting high-quality sound in real time, improve call quality by supporting high-precision voice processing even in the ICS environment of ships, and make it suitable for operational and control environments requiring high response and high-reliability communication.

[0041] According to an embodiment of the present invention, when converting an analog sound source into a digital sound source during a call on a naval vessel, 48,000 samples are collected per second, meaning that the maximum frequency range that can be processed by sampling the analog voice 48,000 times per second is very high, and the expressible range is 32 bits, meaning that since data is processed within a 32-bit range, it can have 4.29 billion values, thus providing high precision. Furthermore, since the data is not compressed when processing the voice, it can be processed with minimal delay. Since it is already a 32-bit Float type, it can be processed with minimal delay during voice mixing, and it has the effect of enabling the playback of high-quality sound sources (e.g., music or SONAR, etc.) or high-quality sound sources. Brief explanation of the drawing

[0042] FIG. 1 is a drawing illustrating a voice call format application device in an integrated communication system for a vessel according to an embodiment of the present invention as a first example. Figure 2 is a diagram illustrating, as an example, 44,100Hz sampling and 32-bit float data transmitted and received between a communication terminal and an analog interconnected terminal in Figure 1. FIG. 3 is a drawing illustrating a second example of a voice call format application device in an integrated communication system for a vessel according to an embodiment of the present invention. FIG. 4 is a diagram illustrating a method for applying a voice call format in an integrated communication system for a vessel according to an embodiment of the present invention. Specific details for implementing the invention

[0043] Below, with reference to the attached drawings, embodiments of the present invention are described in detail so that those skilled in the art can easily implement the invention. However, since the description of the present invention is merely an example for structural or functional explanation, the scope of the rights of the present invention should not be interpreted as being limited by the embodiments described in the text. That is, since the embodiments are subject to various modifications and may take various forms, the scope of the rights of the present invention should be understood to include equivalents capable of realizing the technical concept. Furthermore, the objectives or effects presented in the present invention do not imply that a specific embodiment must include all of them or only such effects; therefore, the scope of the rights of the present invention should not be understood as being limited by them.

[0044] The meaning of the terms described in this invention should be understood as follows.

[0045] Terms such as "first" and "second" are intended to distinguish one component from another, and the scope of rights shall not be limited by these terms. For example, the first component may be named the second component, and similarly, the second component may be named the first component. When a component is referred to as being "connected" to another component, it should be understood that it may be directly connected to that other component, or that there may be other components in between. Conversely, when a component is referred to as being "directly connected" to another component, it should be understood that there are no other components in between. Meanwhile, other expressions describing the relationship between components, such as "between" and "exactly between," or "adjacent to" and "directly adjacent to," shall be interpreted in the same manner.

[0046] A singular expression should be understood to include a plural expression unless the context clearly indicates otherwise, and terms such as "include" or "have" are intended to specify the existence of the set-up features, numbers, steps, actions, components, parts, or combinations thereof, and should be understood not to preclude the existence or addition of one or more other features, numbers, steps, actions, components, parts, or combinations thereof.

[0047] Unless otherwise defined, all terms used herein have the same meaning as generally understood by those skilled in the art to which this invention pertains. Terms defined in commonly used dictionaries should be interpreted as having meanings consistent with the context of the relevant technology and should not be interpreted as having an ideal or overly formal meaning unless explicitly defined in this invention.

[0048] Now, a device and method for applying a voice call format in an integrated communication system for a vessel according to an embodiment of the present invention will be described in detail with reference to the drawings.

[0049] FIG. 1 is a drawing illustrating a voice call format application device in an integrated communication system for a vessel according to an embodiment of the present invention as a first example, and FIG. 2 is a drawing illustrating 44,100 Hz sampling and 32-bit float data transmitted and received between a communication terminal and an analog linked terminal in FIG. 1 as an example.

[0050] Referring to FIGS. 1 and 2, a voice call format application device (100) in an Integrated Communication System (ICS) for a vessel includes an audio mixer controller (110), an audio mixer (120), a plurality of communication terminals (131, 132), an analog interlocking terminal (140), and an analog sound device (150).

[0051] The audio mixer controller (110) performs audio mixing settings to control the transmission and reception of a format-applied audio signal and transmits the set audio mixing information to the audio mixer (120).

[0052] The audio mixer (120) receives a format-applied audio signal transmitted from a communication terminal (131, 132) or an analog linkage terminal (140) according to audio mixing information transmitted from the audio mixer controller (110), mixes it, and transmits the mixed format-applied audio signal to the analog linkage terminal (140) or the communication terminal (131, 132).

[0053] In one embodiment, the audio mixer (120) can use 44,100Hz sampling and 32-bit float data received from a communication terminal (131, 132) or an analog linkage terminal (140) to mix directly without data conversion during the actual mixing operation.

[0054] The communication terminal (131, 132) applies a high-quality voice call format to convert an analog audio signal received through an input means such as an external device or microphone into a format-applied audio signal and transmits it to an audio mixer (120), and receives the format-applied audio signal transmitted from the audio mixer (120), converts it into an analog audio signal, and outputs it through an output means such as a speaker.

[0055] In one embodiment, the communication terminal (131, 132) can perform ICS high-quality voice calls by improving voice quality through the application of a floating-point based 32-bit (Float 32-bit) Pulse Code Modulation (PCM) voice format in packet-based communication as a high-quality voice call format, in order to process high-quality voice signals while minimizing delay during a call.

[0056] In one embodiment, the communication terminal (131, 132) converts an analog audio signal transmitted from an analog sound device (150) into a format-applied audio signal by applying a floating-point-based 32-bit PCM voice format that provides a wider dynamic range (or expression range) and precision than a conventional integer-based voice format (e.g., 16-bit 8000 sample PCM), thereby reducing distortion or loss that may occur during a voice call and enabling high-quality communication close to the original sound.

[0057] In one embodiment, the communication terminal (131, 132) applies a floating-point-based 32-bit PCM voice format as shown in FIG. 2 when processing an analog audio signal, converting it into 44,100 Hz sampling and 32-bit float data, and can transmit it to an audio mixer (120) by sampling 48,000 times per second and generating float data in a 32-bit form.

[0058] In one embodiment, the communication terminal (131, 132) is configured to be linked with a packet-based real-time communication structure in the case of a floating-point-based 32-bit PCM voice format, that is, to be transmitted in conjunction with the voice packet processing flow of the ICS, and is optimized to enable real-time encoding / decoding and streaming in a high-speed network environment, and also transmits by applying a delay correction algorithm in parallel to prepare for packet loss.

[0059] In one embodiment, the communication terminal (131, 132) may be equipped with lightweight processing logic for minimizing delay, and in order to achieve low delay required by ICS characteristics while maintaining high sound quality using said lightweight processing logic, the format conversion and compression steps are minimized and applied during voice format processing, thereby ensuring real-time responsiveness while maintaining high-resolution sound quality.

[0060] The analog linked terminal (140) applies a high-quality voice call format to convert an analog audio signal transmitted from an analog sound device (150) into a format-applied audio signal and transmits it to an audio mixer (120), and receives a format-applied audio signal transmitted from an audio mixer (120), converts it into an analog audio signal, and transmits it to an analog sound device (150).

[0061] In one embodiment, the analog interlocking terminal (140) applies a floating-point-based 32-bit PCM voice format as shown in FIG. 2 when processing analog audio signals, converting them into 44,100 Hz sampling and 32-bit float data, and can transmit to the audio mixer (120) the float data generated in a 32-bit form by sampling 48,000 times per second.

[0062] In one embodiment, the analog interlocking terminal (140) is configured to be linked with a packet-based real-time communication structure in the case of a floating-point-based 32-bit PCM voice format, that is, to be transmitted in conjunction with the voice packet processing flow of the ICS, and is optimized to enable real-time encoding / decoding and streaming in a high-speed network environment, and also transmits by applying a delay correction algorithm in parallel to prepare for packet loss.

[0063] In one embodiment, the analog interlocking terminal (140) may be equipped with a lightweight processing logic for minimizing delay, and in order to achieve low delay required by ICS characteristics while maintaining high sound quality using the lightweight processing logic, the format conversion and compression steps are minimized and applied during voice format processing, thereby ensuring real-time responsiveness while maintaining high-resolution sound quality.

[0064] The analog sound device (150) receives an analog audio signal through an input means and transmits it to an analog linked terminal (140), and outputs the analog audio signal transmitted from the analog linked terminal (140) through an output means.

[0065] FIG. 3 is a drawing illustrating a second example of a voice call format application device in an integrated communication system for a vessel according to an embodiment of the present invention.

[0066] The voice call format application device (100) in the integrated communication system for traps further includes a network switch (160) and a SONAR device (170) (or a music device (not shown in the drawing for convenience of explanation)) so that, as shown in FIG. 3 as another example, not only audio but also sonar (sound navigation ranging; SONAR) or music, which were previously impossible to link due to quality issues, can be monitored remotely via the network due to improved voice call quality.

[0067] In one embodiment, the audio mixer controller (110) performs sonar (or music) mixing settings to control the transmission and reception of a format-applied sonar signal (or music data) and transmits the set sonar (or music) mixing information to the audio mixer (120).

[0068] In one embodiment, the audio mixer (120) receives a format-applied sonar signal (or music data) transmitted from a network switch (160) or an analog-linked terminal (140) according to sonar (or music) mixing information transmitted from the audio mixer controller (110), mixes it, and transmits the mixed format-applied sonar signal (or music data) to a communication terminal (131) via the analog-linked terminal (140) or the network switch (160).

[0069] In one embodiment, the audio mixer (120) can be used to mix directly without data conversion during actual mixing operations by using 44,100Hz sampling and 32-bit float data received from an intermediate network switch (160) or an analog interconnected terminal (140).

[0070] In one embodiment, the communication terminal (131, 132) enables monitoring of the analog sonar signal (or music data of the music device) of the sonar device (170) from a remote location at a distance.

[0071] In one embodiment, the communication terminal (131, 132) applies a high-quality voice call format to convert an analog sonar signal (or music data) received through an input means into a format-applied sonar signal (or music data) and transmits it to an audio mixer (120) via a network switch (160), and receives a format-applied sonar signal (or music data) transmitted from the network switch (160), converts it into an analog sonar signal (or music data), and outputs it through an output means.

[0072] In one embodiment, the communication terminal (131, 132) can convert the analog sonar signal (or music data) into 44,100Hz sampling and 32-bit float data by applying a floating-point based 32-bit PCM voice format as shown in the example in FIG. 3 when processing the analog sonar signal (or music data).

[0073] The analog linked terminal (140) applies a high-quality voice call format to convert an analog sonar signal (or music data transmitted from a music device) transmitted from a sonar device (170) into a format-applied sonar signal (or music data) and transmits it to an audio mixer (120), and receives a format-applied sonar signal (or music data) transmitted from an audio mixer (120), converts it into an analog sonar signal (or music data), and transmits it to a sonar device (170) (or music device).

[0074] In one embodiment, the analog linked terminal (140) can convert the analog sonar signal (or music data) into 44,100Hz sampling and 32-bit float data by applying a floating-point based 32-bit PCM voice format as shown in the example in FIG. 3 when processing the analog sonar signal (or music data).

[0075] A network switch (160) is formed between an audio mixer (120) and a plurality of communication terminals (131, 132) and switches a format-applied sonar signal (or music data) between the audio mixer (120) and the plurality of communication terminals (131, 132).

[0076] The sonar device (170) (or music device) receives an analog sonar signal (or music data) through an input means and transmits it to an analog linked terminal (140), and outputs the analog sonar signal (or music data) transmitted from the analog linked terminal (140) through an output means.

[0077] The voice call format application device (100) in the integrated communication system for ships having the configuration described above is implemented to improve voice quality by applying a high-quality voice call format in the environment of the integrated communication system for ships to minimize delay during a call while processing high-quality voice signals, thereby enabling improved voice resolution and minimized distortion compared to the existing one by applying a float 32-bit 48,000 sample PCM, and enabling the implementation of low-latency calls that can transmit high-quality sound in real time, and can improve call quality by supporting high-precision voice processing even in the ICS environment of ships, and can be made suitable for operational and control environments that require high response and high-reliability communication.

[0078] The voice call format application device (100) in the integrated communication system for naval vessels having the configuration described above collects 48,000 samples per second to convert analog sound sources into digital sound sources during calls on naval vessels, that is, by sampling analog voice 48,000 times per second, the maximum frequency range that can be processed is very high, and the expressible range is 32 bits, that is, since data is processed in a 32-bit range, it can have 4.29 billion values, so the precision is high, and since the data is not compressed when processing voice, it can be processed with minimal delay, and since it is already a 32-bit float type, it can be processed with minimal delay when mixing voice, and it is possible to play high-quality sound sources (e.g., music or sonar, etc.) or high-quality sound sources.

[0079] FIG. 4 is a diagram illustrating a method for applying a voice call format in an integrated communication system for a vessel according to an embodiment of the present invention.

[0080] Referring to FIG. 4, the audio mixer controller (110) performs audio mixing settings to control the transmission and reception of a format-applied audio signal and transmits the set audio mixing information to the audio mixer (120) (S401).

[0081] When audio mixing information is transmitted in the above-described step S401, the audio mixer (120) receives a format-applied audio signal transmitted from a communication terminal (131, 132) or an analog linkage terminal (140) according to the audio mixing information transmitted from the audio mixer controller (110), mixes it, and transmits the mixed format-applied audio signal to the analog linkage terminal (140) or the communication terminal (131, 132) (S402).

[0082] In transmitting the format-applied audio signal in the above-described step S402, the audio mixer (120) can use 44,100Hz sampling and 32-bit float data received from the communication terminal (131, 132) or the analog linkage terminal (140) to perform mixing immediately without data conversion during the actual mixing operation.

[0083] When transmitting a format-applied audio signal in the above-described step S402, the communication terminal (131, 132) applies a high-quality voice call format and converts an analog audio signal received through an input means such as an external device or microphone into a format-applied audio signal and transmits it to an audio mixer (120), and receives the format-applied audio signal transmitted from the audio mixer (120), converts it into an analog audio signal, and outputs it through an output means such as a speaker (S403).

[0084] In outputting an analog audio signal on the terminal side in the above-described step S403, the communication terminal (131, 132) applies a floating-point based 32-bit (Float 32-bit) PCM (Pulse Code Modulation) voice format in packet-based communication as a high-quality voice call format to process a high-quality voice signal while minimizing delay during a call, thereby improving voice quality and enabling ICS high-quality voice calls.

[0085] In outputting an analog audio signal on the terminal side in the above-described step S403, the communication terminal (131, 132) applies a floating-point based 32-bit PCM voice format that provides a wider dynamic range (or expression range) and precision than a conventional integer-based voice format (e.g., 16-bit 8000 sample PCM) and converts the analog audio signal transmitted from the analog sound device (150) into a format-applied audio signal, thereby reducing distortion or loss that may occur during a voice call and enabling high-quality communication close to the original sound.

[0086] In outputting the analog audio signal on the terminal side in the above-described step S403, the communication terminal (131, 132) applies a floating-point-based 32-bit PCM voice format as shown in the example in FIG. 2 when processing the analog audio signal, converting it into 44,100 Hz sampling and 32-bit float data, and can transmit it to the audio mixer (120) by sampling 48,000 times per second and generating float data in a 32-bit form.

[0087] In outputting an analog audio signal on the terminal side in the above-described step S403, the communication terminal (131, 132) is configured to be linked with a packet-based real-time communication structure in the case of a floating-point-based 32-bit PCM voice format, that is, transmitted in conjunction with the voice packet processing flow of the ICS, and is optimized to enable real-time encoding / decoding and streaming in a high-speed network environment, and also transmits by applying a delay correction algorithm to prepare for packet loss in parallel.

[0088] In the above-described step S403, when outputting an analog audio signal on the terminal side, the communication terminal (131, 132) equipped with lightweight processing logic for minimizing delay utilizes the said lightweight processing logic to minimize the format conversion and compression steps during voice format processing in order to achieve low delay required by ICS characteristics while maintaining high sound quality, thereby ensuring real-time responsiveness while maintaining high-resolution sound quality.

[0089] In step S403 described above, while the analog audio signal from the terminal side is output, the analog linked terminal (140) applies a high-quality voice call format to convert the analog audio signal transmitted from the analog sound device (150) into a format-applied audio signal and transmits it to the audio mixer (120), and receives the format-applied audio signal transmitted from the audio mixer (120), converts it into an analog audio signal, and transmits it to the analog sound device (150) (S403).

[0090] In transmitting an analog audio signal in the above-described step S403, the analog interconnect terminal (140) applies a floating-point-based 32-bit PCM voice format as shown in the example in FIG. 2 when processing the analog audio signal, converting it into 44,100 Hz sampling and 32-bit float data, and can transmit it to the audio mixer (120) by sampling 48,000 times per second and generating float data in a 32-bit form.

[0091] In transmitting an analog audio signal in the above-described step S403, the analog interconnected terminal (140) is configured to be linked with a packet-based real-time communication structure in the case of a floating-point-based 32-bit PCM voice format, that is, transmitted in conjunction with the voice packet processing flow of the ICS, and is optimized to enable real-time encoding / decoding and streaming in a high-speed network environment, and also transmits by applying a delay correction algorithm to prepare for packet loss in parallel.

[0092] In transmitting an analog audio signal in the above-described step S403, the analog interconnect terminal (140) may be equipped with a lightweight processing logic for minimizing delay, and in order to achieve low delay required by ICS characteristics while maintaining high sound quality using the said lightweight processing logic, the format conversion and compression steps are minimized and applied during voice format processing, thereby ensuring real-time responsiveness while maintaining high-resolution sound quality.

[0093] When transmitting an analog audio signal in step S403 described above, the analog sound device (150) receives the analog audio signal through an input means and transmits it to an analog linked terminal (140), and outputs the analog audio signal transmitted from the analog linked terminal (140) through an output means (S404).

[0094] For the above, embodiments of the present invention are not limited to implementation only through the devices and / or operating methods described above, but may also be implemented through a program for realizing functions corresponding to the configuration of the embodiments of the present invention, a recording medium on which such program is recorded, etc. Such implementation can be easily achieved by a person skilled in the art to which the present invention pertains from the description of the embodiments explained above. Although embodiments of the present invention have been described in detail above, the scope of the present invention is not limited thereto, and various modifications and improvements by those skilled in the art using the basic concepts of the present invention as defined in the following claims also fall within the scope of the present invention. Explanation of the symbols

[0095] 100: Device for applying voice call formats in an integrated communication system for traps 110: Audio Mixer Controller 120: Audio Mixer 130: Communication terminal 140: Analog interconnected terminal 150: Analog Sound Devices 160: Network Switch 170: Sonar Device

Claims

Claim 1 An audio mixer controller that performs audio mixing settings to control the transmission and reception of a format-applied audio signal and transmits the configured audio mixing information; an audio mixer that receives a format-applied audio signal according to the audio mixing information transmitted from the audio mixer controller, mixes it, and transmits the mixed format-applied audio signal; a communication terminal that converts an analog audio signal received through an input means into a format-applied audio signal by applying a voice call format and transmits it to the audio mixer, and receives the format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and outputs it through an output means; and an analog interlocking terminal that receives an analog audio signal by applying a voice call format, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives the format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and transmits it. A voice call format application device in an integrated communication system for a vessel, characterized by further including: an analog sound device that receives an analog audio signal through an input means and transmits it to the analog linked terminal, and outputs the analog audio signal transmitted from the analog linked terminal through an output means; the audio mixer mixes directly without data conversion during a mixing operation using 44,100Hz sampling and 32-bit float data received from the communication terminal or the analog linked terminal; a network switch formed between the audio mixer and the communication terminal to switch a format-applied sonar signal or music data between the audio mixer and the communication terminal; and a sonar device that receives an analog sonar signal through an input means and transmits it to the analog linked terminal, and outputs the analog sonar signal transmitted from the analog linked terminal through an output means. Claim 2 delete Claim 3 A voice call format application device in an integrated communication system for a vessel, characterized in that, in claim 1, the communication terminal applies a floating-point-based 32-bit PCM voice format in packet-based communication as the voice call format. Claim 4 A voice call format application device in an integrated communication system for a vessel, characterized in that, in paragraph 3, the communication terminal applies the floating-point-based 32-bit PCM voice format during analog audio signal processing and converts it into 44,100Hz sampling and 32-bit float data. Claim 5 A step in which an audio mixer controller performs audio mixing settings to control the transmission and reception of a format-applied audio signal and transmits the configured audio mixing information; a step in which an audio mixer receives a format-applied audio signal according to the audio mixing information transmitted from the audio mixer controller, mixes it, and transmits the mixed format-applied audio signal; a step in which a communication terminal applies a voice call format to an analog audio signal received through an input means, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives the format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and outputs it through an output means; a step in which an analog-linked terminal applies a voice call format to an analog audio signal, receives it, converts it into a format-applied audio signal, and transmits it to the audio mixer, and receives the format-applied audio signal transmitted from the audio mixer, converts it into an analog audio signal, and transmits it. A method for applying a voice call format in an integrated communication system for a vessel, characterized by further including: an analog sound device receiving an analog audio signal through an input means and transmitting it to the analog linked terminal, and outputting the analog audio signal transmitted from the analog linked terminal through an output means; the audio mixer mixing immediately without data conversion during the mixing operation using 44,100Hz sampling and 32-bit float data received from the communication terminal or the analog linked terminal; a network switch formed between the audio mixer and the communication terminal to switch a format-applied sonar signal or music data between the audio mixer and the communication terminal; and a sonar device receiving an analog sonar signal through an input means and transmitting it to the analog linked terminal, and outputting the analog sonar signal transmitted from the analog linked terminal through an output means.

Images