• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.5
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• pp.25-33
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• 2010

Multi-media service has been changed into user based audio services, which service supports actively user's preference and interaction with the users. In the market, multi-media products which can support the highest audio-quality by using lossless audio technology have been released and object audio music which user can select the objects has been serviced. In this paper, we design user's preference information based object audio file format and audio scene description for storage and transmission media. The designed file format is designed based on MPEG-4 file format because high-quality audio codecs in MPEG-4 audio can be easily used and the track of file format can be flexibly controlled depend on the number of the instrument in music. The encoded audio data of each objects and encoded audio scene description by binary encoding that has independent track are packed in a file. The scene description for storage media is consist of full and object scene description, the scene description for transmission media has an essential description for object audio operation and a specific description for real audio sound. The designed file format based simulator is developed and it generates an object audio file with several scene descriptions. Also, the real audio sound is serviced by the interaction with user and the unpacked scene description.

• Proceedings of the IEEK Conference
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• 2002.06e
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• pp.55-58
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• 2002

This paper describes an RF module for 2.4GHz wireless audio/video transmission. The pre-processed baseband input signals are FM-modulated using a VCO and then transmitted through an antenna after RF filtering. The designed circuits are implemented using a Teflon board of which the size is 52mm${\times}$62mm. The measured maximum output signal levels are around -3dBm and the harmonics are less than -450dBc. The manufactured module consumes 130mA from a 8V supply.

• Journal of Korea Multimedia Society
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• v.4 no.4
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• pp.297-305
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• 2001

Recently advances in high-speed networks and multimedia computer technologies allow new types of multimedia applications to manipulate large volumes of multimedia data. However, in the real time and/or the heterogeneous data transmissions, there are many difficulties such as network transmission delay, the implementation difficulties, and so on. To solve these problems, in this paper, we extend the method of the multimedia service design which is proposed by OMG. To do this, we suggest an efficient real time audio/video stream framework, called Smart Explorer, based un CORBA and JMF Java Media API. And we separate the transmission path of control data from that of media data and use RTP/RTCP protocol for efficient real time audio/video transmission. Also we show the appropriate implementation of the audio/video stream system based on our suggested framework Smart Explorer. In the future, we expect our audio/video stream system to be applied to the real time communication software such as broadcasting, distance learning, and video conferencing.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.12
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• pp.2165-2170
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• 2008

In this parer, it is implemented a Local distribution Audio System, based on AoIP(Audio over Internet Protocol) of a part of TCP/IP Network which belongs to Internet transmission technology. The system is controlled based on SNMP(Simple Network Management Protocol) and it is transferred to UDP as packet after changing from Analog audio sources to Digital audio sources. The implemented Local distribution Audio System have presented practical possibilities in PA system transmitting various audio sources to several areas, dispersedly and using multichannel audio like Home theaters in the limelight, recently.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.357-368
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• 2016

In DAB+, HE-AAC v2 codec is applied for the fixed rate transmission of audio stream. In case that xHE-AAC codec including USAC, a more efficiency is expected when the variable frame is used in a given same bandwidth compared to the fixed frame transmission. For this to be realized, audio streams need to be multiplexed in a sub-channel before transmission, then a method is required to identify the border of each audio frames. In this paper, the toggled sync byte and additional identification field being sequentially placed between AU borders are proposed in order to deal with the AU border identification. In addition, the Reed-Solomon based error correction code which is compliant to DAB+ is proposed.

• Journal of Internet Computing and Services
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• v.16 no.2
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• pp.1-6
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• 2015

In this paper, we designed an embedded audio video bridging (AVB) platform based on IEEE 802.1BA for real-time multimedia transmission in smart-car, smart-home, smart-theater, and then evaluated a performance of the implemented platform by analysis of IEEE 802.1AS (time synchronization protocol) and IEEE 802.1Qat (stream reservation protocol). Especially, the AVB Layer-2 protocol of MRP(Multiple Registration Protocol), MMAP(Multicast Address Acquisition Protocol), IEEE 1722, 1722.1 etc. was and implemented by linux based operating system. It is shown by interoperability tests with commercial products that the implemented platform transmits real-time multichannel AV data over AVB networks for Multichannel Multimedia Transmission.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2017.05a
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• pp.656-659
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• 2017

In this paper, we propose a digital method for transmitting audio signals using LED visible light communication system. In the proposed method, we compare the method for transmitting audio signal in analog signal and the method for transmitting by digital signal. When amplifying the audio sound and transmitting the analog signal using the LED visible light communication, attenuation corresponding to the transmission distance occurs, and there is a disadvantage that it is noisy. In order to overcome this, we propose a method for transmitting digital audio signals. The proposed method has the advantage of reducing the influence of noise, but it turned out that it is affected much by the LED blinking speed. Various methods to overcome this need to be continuously studied.

• Proceedings of the KIEE Conference
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• 1998.07g
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• pp.2262-2264
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• 1998

This paper propose a new method for the discrete signal : Discrete Wavelet Transform(DWT). This paper is a brief introduction to the DWT and applies the DWT coding for the audio data as an example. We can have a number of hint about the compression algorithm of multimedia resources and the high performance of transmission and storage.

• Journal of Internet Computing and Services
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• v.21 no.4
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• pp.51-57
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• 2020

Currently, High-efficient video coding (HEVC) has become the most promising video coding technology. However, the implementation of HEVC in video streaming systems is restricted by factors such as cost, design complexity, and compatibility with existing systems. While HEVC is considering deploying to various systems with different reached methods, H264/AVC can be one of the best choices for current video streaming systems. This paper presents an adaptive method for manipulating video streams using video coding on an integrated circuit (IC) designed with a private network processor. The proposed system allows to transfer multimedia data from cameras or other video sources to client. For this work, a series of video or audio packages from the video source are forwarded to the designed IC via HDMI cable, called Tx transmitter. The Tx processes input data into a real-time stream using its own protocol according to the Real-Time Transmission Protocol for both video and audio, then Tx transmits output packages to the video client though internet. The client includes hardware or software video/audio decoders to decode the received packages. Tx uses H264/AVC or HEVC video coding to encode video data, and its audio coding is PCM format. By handling the message exchanges between Tx and the client, the transmitted session can be set up quickly. Output results show that transmission's throughput can be achieved about 50 Mbps with approximately 80 msec latency.

• The KIPS Transactions:PartA
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• v.15A no.6
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• pp.317-324
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• 2008

In this paper, we propose a system which easily controls the existing serially connected multi-channel speakers in a general personal computer by using a USB(Universal Serial Bus) interface. The personal computer as a host of the USB interface analyzes a sound source and sends audio data in a real-time fashion by the use of the isochronous transmission, one of four transmission methods provided by the USB interface. In addition, a channel is assigned by means of the bulk transmission, one of four transmission methods provided by the USB interface. Transmitted data from the USB host are sent to each speaker through compression and packet generation process. Each speaker detects corresponding digital data and regenerates audio signals through DAC(Digital-to-Analog Converter). A user can easily select a sound source file and a channel by the use of a GUI environment in a personal computer.