• IEMEK Journal of Embedded Systems and Applications
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• v.6 no.3
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• pp.174-182
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• 2011

Conventional home audio system (e.g. AV Receiver, CD Player etc) has a various functionality of audio play, channel mixing, but the remote controller of these audio players is too complex, difficult for user to manage them effectively. Users want to use these functionalities with more easy, comprehensible way. In this study, "intelligent real-sound presentation technology" that support high quality, realistic audio and the "design of complex information and controller of real sound using intelligent real sound play and control interface" will be introduced. So user can actively, realistically enjoy and play real sound based on user's preference, emotion and circumstance, instead of user's passive service.

• Journal of KIISE:Information Networking
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• v.33 no.2
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• pp.139-148
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• 2006

Recently with the advancement of local wireless communication technologies, digital trends of audio contents and devices are paving the way for new experiences to network-based audio streaming services. In this paper, we present the Wireless Audio Sharing (WASH) technology over Bluetooth and Wireless Local Area Network (WLAN). WASH system provides an audio sharing mechanism over multiple users and also provides stereo audio streaming between Bluetooth-enabled devices and Universal Plug and Play (UPnP) media devices connected to wired/wireless LANs. To achieve these, we extended the Bluetooth audio distribution profile and combined the extended profile with the UPnP AV architecture. With the implementation of this WASH technology, we show some experimental results of the stereo audio streaming in a real environment.

• ETRI Journal
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• v.38 no.5
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• pp.829-837
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• 2016

In this paper, we introduce extension tools for MPEG Surround, which were recently adopted as MPEG-H 3D Audio tools by the ISO/MPEG standardization group. MPEG-H 3D Audio is a next-generation technology for representing spatial audio in an immersive manner. However, considerably large numbers of input signals can degrade the compression performance during a low bitrate operation. The proposed extension of MPEG Surround was basically designed based on the original MPEG Surround technology, where the limitations of MPEG Surround were revised by adopting a new coding structure. The proposed MPEG-H 3D Audio technologies will play a pivotal role in dramatically improving the sound quality during a lower bitrate operation.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.689-691
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• 2011

Due to the rapid development of Information and communication, the technology of multimedia presentation technology is evolving into the service that user can actively, realistically enjoy and play based on user's preference and taste not only for User's passive service. Especially, the industry related the realistic multimedia service that supports targeting Human emotion with the property of Human hearing is expected to be formed of the high value-added premium market. Audio technology is affected on human's emotion and the viewing environment around than video technology. Also the audio technology compared to video technology is a research part that appeals to human emotion and emphasize on psychological aspects. With this viewpoint, the development of intelligent and realistic audio technology needs highly specialty. In this study, "intelligent real-sound presentation technology" that support high quality and realistic audio and the "core technologies" that are composing of this will be introduced.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.15 no.12
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• pp.2705-2711
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• 2011

The recent development in multimedia technology such as audio technology needs high quality audio system. Especially, Real Audio Technology is to be developed to play realistic sound. To meet this demands, researches on 3-Dimensional Audio which provides realistic audio effect in virtual reality and augmented reality are conducted. In this paper, how to provide realistic audio effect by using better audio technologies in augmented reality was investigated. In the study, the movements of the 3-Dimensional model on the markers were used to provide the sense of reality in virtual and real world. Namely, the sound was modified according to the movement of the model. The change in distance and angle of the model affected the sound volume and the pitch.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.49-54
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• 2009

Synchronization between media is an important aspect in the design of multimedia streaming system. This paper proposes a precise media synchronization mechanism for digital video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio streams. With the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• ETRI Journal
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• v.33 no.6
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• pp.945-948
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• 2011

Object-based audio coding can provide new music applications with interactivity. To efficiently compress a lot of target audio objects, a subband-based parametric coding scheme has been adopted for MPEG spatial audio object coding. In this letter, the time-frequency (T/F) subband analysis structure is investigated. A reconfigured T/F structure is also proposed to enhance the generating performance of sound scenes such as 'karaoke' and 'solo' play in interactive music scenarios. From the experimental results, it was confirmed that the proposed scheme remarkably improves the SNR and sound quality.

• Journal of Korea Multimedia Society
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• v.12 no.5
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• pp.653-663
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• 2009

Synchronization between media is an important aspect in the design of multimedia communication-system. This paper proposes a precise media synchronization mechanism for video and audio transport over IP networks. To support synchronization between video and audio bitstreams transported over IP networks, RTP/RTCP protocol suite is usually employed. To provide a precise mechanism for media synchronization between video and audio, we suggest an efficient media synchronization algorithm based on NPT (Normal Play Time) which can be derivable from the timestamp information in the header part of RTP packet generated for the transport of video and audio. In the proposed method, we do not need to send and process any RTCP SR (sender report) packet which is required for conventional media synchronization scheme, and accordingly could reduce the number of required UDP ports and the amount of control traffic injected into the network.

• Journal of Korea Society of Industrial Information Systems
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• v.13 no.5
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• pp.117-123
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• 2008

In this paper, we develop MPEG-4 AAC streaming player on embedded Linux processor such as mobile terminals. Moreover we show the experimental results that the player preforms the decoding processes of MPEG-4 AAC data effectively. MPEG-4 AAC technology supports a wide range encoding rates and high sound quality so it is appropriate to adopt various applications. In particular, the need in the development of the application of audio data increases according to significantly increase in devices used in mobile environments, such as cell phones and PDAs. In this environment, it is necessary to optimize the decoding processes to the ability of the terminal hardware in order to play audio data without delays. We also implement the decoding module to optimize the processor capabilities and make the player to decode and play streaming audio data from streaming server.

• Proceedings of the KIEE Conference
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• 1989.07a
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• pp.486-490
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• 1989

In this paper, a program controlling an auto-audio media - cassette deck - by a 16 bit personal computer is studied in order to execute audio and visual learning in CAI. The results of this study are as follows. 1. Audio and visual learning is executed efficiently in CAI. 2. Access rate of voice information to text/image information is about 98% and 60% in "play" and "fast forward" respectively. 3. In "fast forward", quality of a cassette tape affects voice information access rate in propotion to motor driving speed. 4. Synchronizing signal may be mistaken by defects of tape itself.