• 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.

• 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.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.283-286
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• 2004

Due to advances in electronics very high quality audio reproduction is today possible. But the listening environment causes deviation of the audio system from the expected behavior. Firstly the listening Room significantly changes the audio signal frequencies and their phase. Secondly the position of the user in the room affects the perceived sound. With existing DSP technology it is possible to adequately correct these effects. In our work we developed a room correction system, correcting up to 7.1 channels using dual Motorola 56367 fixed point DSP's, implementing position dependent room effects measurement, real time compensation filter design and equalization filtering procedures.

• ETRI Journal
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• v.28 no.4
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• pp.509-512
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• 2006

In a real environment, sound recordings are commonly distorted by channel and background noise, and the performance of audio identification is mainly degraded by them. Recently, Philips introduced a robust and efficient audio fingerprinting scheme applying a differential (high-pass filtering) to the frequency-time sequence of the perceptual filter-bank energies. In practice, however, the robustness of the audio fingerprinting scheme is still important in a real environment. In this letter, we introduce alternatives to the frequency-temporal filtering combination for an extension method of Philips' audio fingerprinting scheme to achieve robustness to channel and background noise under the conditions of a real situation. Our experimental results show that the proposed filtering combination improves noise robustness in audio identification.

• Proceedings of the IEEK Conference
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• 2002.07b
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• pp.872-875
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• 2002

A digital watermarking technique applying psychoacoustic model for audio signal is proposed in this paper. In the watermarking scheme, the pseudo-random bit stream used as a watermark signal is embedded into the audio signal in both speech and music. The strength of the embedded signal is subject to the human auditory system in such a way that the disturbances on host audio signal are beyond the sensing of human ears. The experimental results show that the quality of the watermarked audio signal, in term of signal to noise ratio, can be improved up to 3.2 dB.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2018.06a
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• pp.158-160
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• 2018

최근 다양한 국제 방송표준에서 차세대 오디오 코덱의 하나로 MPEG-H 3DA(3D Audio)가 채택되었으며, 이를 활용한 몰입형 오디오 서비스들이 개발되고 있다. 이러한 몰입형 서비스를 원활히 제공하기 위해서는 표준에 정의된 기술을 구현한 제품간의 상호호환성 검증이 필수적으로 추진되어야 하며, 이를 위해 개발된 MPEG-H TV Audio System 인증 프로그램에 대해 대상 제품과 시험서비스의 구조에 대해 설명하고 있다.

• Proceedings of the Korea Information Processing Society Conference
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• 2013.11a
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• pp.1450-1451
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• 2013

본 논문에서는 음향 확성 환경과 음향신호의 입 출력 조건에 최적화된 음향 시스템을 실시간으로 생성 및 적용 가능한 음향 신호처리용 시스템으로서, 가변형 필터뱅크 기술 및 상황 적응적 필터 재조합 재배열 기술을 음향 신호처리용 DSP에 적용함으로서 Pro-Audio 장비, 방송 음향장비, 산업 음향장비와 같은 다양한 음향관련 장비에서 고품질 음향 서비스를 제공하기위한 핵심 기술인 가변형 필터뱅크 기반 고품질 음향 DSP 시스템 개발을 제안한다.

• Journal of Broadcast Engineering
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• v.12 no.3
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• pp.222-229
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• 2007

According to the advancement of multimedia compression technologies, high quality multi-media services are easily found in common life. Along with this situation, 5.1-channel audio service also has expanded the application area to home theater system and car theater system and consumer can easily take a chance to experience the feeling of 5.1-channel audio. On the other hand, terrestrial DMB service has been launched in Korea from Dec. 2005 as a handhold multi-media broadcasting service. However, multi-channel audio was not considered due to the insufficiency of bandwidth and the handhold usage. Lately, MPEG is standardizing high efficiency multi-channel audio compression technology for handheld broadcasting service, and several trial for application is introduced in Europe. In this paper, we would like to explain multi-channel audio compression technology, describe the implementation of the verification system for the multi-channel audio service over T-DMB and investigate the possibility of further realization of the service.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.10a
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• pp.619-621
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• 2013

This paper is proposing a real-time audio processing system for TCP/IP with server-client. The server sends the audio data packet which is the same size each time while playing the audio data. And the client plays the received audio data from the server. In general, The receiving speed of audio data packet is faster than processing the audio data. So, the unstable playback is occurred when playing the received audio data at the moment. In order to overcome this problem, the double buffering method is proposed.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1415-1419
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• 2008

In this paper, we investigate various Audio Transmission Systems to implement Audio Transmission System based on AoIP of Internet transmission technology TCP/IP Network and we design and implement a Audio transmission system based on AoIP by adopting the most efficient one. The implemented system can be applied for various professional audio systems with large-scale audio distribution network as well as small-scale PA systems. Moreover, this can be applied in various fields which need audio transmission.