• Broadcasting and Media Magazine
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• v.14 no.2
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• pp.75-87
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• 2009

• Information and Communications Magazine
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• v.23 no.7
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• pp.85-96
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• 2006

• The Journal of the Acoustical Society of Korea
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• v.31 no.6
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• pp.410-418
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• 2012

Parametric stereo(PS) and MPEG surround(MPS) are major spatial audio coding(SAC) tools. In this paper, the problem of the inter-channel correlation(ICC) synthesis in the conventional SAC is analyzed. Conventional methods assume that ambient components mixed to two output channels are anti-phased, while the primary components are assumed to be in-phased. This assumption can cause excessive ambient mixing for a negative-valued ICC. As a remedy to this problem, we propose a new ICC synthesis method based on an assumption that the primary components are anti-phased each other for a negative ICC. The proposed method is also applied to the approximation which works in practice. The performance of the proposed method was evaluated by computer simulations and the subjective listening tests verified that the proposed method is effective in not only headphones but also loudspeakers playback.

• Journal of the Institute of Convergence Signal Processing
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• v.4 no.3
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• pp.7-14
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• 2003

MPEG-1 audio standard is the algorithm for the compression of high-qualify digital audio signals. The standard dictates the functions of encoder and decoder pair, and includes three different layers as the complexity and the performance of the encoder and decoder. In this paper, we implemented the real-time system of MPEG-1 audio layer III decoder(MP3) with the TMS320C541 fixed point DSP chip. MP3 algorithm uses psycho-acoustic characteristic of human hearing system, and it reduces the amount of data with eliminating the signals hard to be heard to the hearing system of human being. It is difficult to implement MP3 decoder with fixed Point DSP because of it's broad dynamic range. We implemented realtime system with fixed DSP chip by using weighted look-up tables to reduce the amount of calculation and solve the problem of broad dynamic range.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.716-719
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• 1999

In this Paper, we Present a method for embedding digital watermarks into digital audio signals. The watermarking must be imperceptible and should be robust to attacks, such as filtering and compression etc. In our method, we adaptively embedded the watermarks changing the scale factor using the spread spectrum and MPEG-2 AAC psychoacoustic model.

• Proceedings of the Korean Information Science Society Conference
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• 2002.04b
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• pp.595-597
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• 2002

MPEG그룹은 미디어 객체를 시간과 3차원 공간상에 배치하여, 멀티미디어 컨텐츠를 표현하고, 사용자의 상호작용을 가능하게 하는 MFEG-4 System물 표준화하였으며, 다양한 전송계층 (File, TCF/IP, Broadcasting등)을 통합하여 처리할 수 있는 MPEG-4 System의 Multimedia Interface Framework)[3] 전송규약을 표준화하였다. 이 논문에서는 MPEG-4 System을 이용하여 비디오, 오디오, 그랙픽과 같은 다양한 미디어들을 결합하해주고 사용자와의 상호작용에 의한 실시간 표현을 가능하게 해주며 사용자가 MPEG-4 Sytem에 대해서 잘 알고 있지 않더라도 손쉽게 MPEG-4미디어 파일을 만들 수 있도록 해주는 저작툴을 제안한다 [1,3,5]

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2005.11a
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• pp.233-235
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• 2005

멀티미디어 콘텐츠를 저작, 생성, 전달을 통해 최종 소비에 이르기까지 콘텐츠를 보호하고 안전하게 다루기 위해 MPEG-21 IPMP(Intellectual Property Management and Protection)가 표준을 정의하고 있으며 현재 FCD(Final Committee Draft) 단계까지 표준화가 이루어졌다. 우리는 MPEG-21 IPMP에서 규정한 FCD규격에 따라 멀티미디어 콘텐츠와 그에 대한 저작권을 보호 관리하는 시스템을 개발하였다. 제안한 MPEG-21 기반 REL및 IPMP 보호 관리 시스템은 크게 보호 관리 메타데이터 생성부, 통합 제어 관리부, 디지털 아이템 암호화부, 네트워크 인터페이스부의 4부분으로 구성된다. 이 시스템은 먼저 라이센스 서버를 통해 MPEG-21 REL(Rights Expression Language) 규격에 맞는 라이센스 정보가 담긴 문서를 파싱하는 기능을 가지며 오디오 또는 비디오 워터마크가 적용된 멀티미디어콘텐츠에 대해 MPEG-21 IPMP 기반의 메타데이터를 생성하고 보호 관련 REL및 IPMP 메타데이터와 멀티미디어 콘텐츠를 Remuxing 하여 전송되며 멀티미디어 콘텐츠 및 하나의 패키징된 DID는 암호화하여 최종 소비자에게 전송한다.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• The Journal of the Acoustical Society of Korea
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• v.14 no.2E
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• pp.91-97
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• 1995

This paper describes the implementation of a multi-channel audio codec for HETV. This codec has the features of the 3/2-stereo plus low frequency enhancement, downward compatibility with the smaller number of channels, backward compatibility with the existing 2/0-stereo system(MPEG-1 audio), and multilingual capability. The encoder of this codec consists of 6-channel analog audio input part with the sampling rate of 48 kHz, 4-channel digital audio input part and three TMS320C40 /DSPs. The encoder implements multi-channel audio compression using a human perceptual psychoacoustic model, and has the bit rate reduction to 384 kbit/s without impairment of subjective quality. The decoder consists of 6-channel analog audio output part, 4-channel digital audio output part, and two TMS320C40 DSPs for a decoding procedure. The decoder analyzes the bit stream received with bit rate of 384 kbit/s from the encoder and reproduces the multi-channel audio signals for analog and digital outputs. The multi-processing of this audio codec using multiple DSPs is ensured by high speed transfer of date between DSPs through coordinating communication port activities with DMA coprocessors. Finally, some technical considerations are suggested to realize the problem of real-time operation, which are found out through the implementation of this codec using the MPEG-2 layer II sudio coding algorithm and the use of the hardware architecture with commercial multiple DSPs.

• Journal of KIISE:Software and Applications
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• v.33 no.2
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• pp.220-230
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• 2006

As the computing technique and mobile devices get developed, the demand of multimedia contents for mobile environment has been increased. The multimedia contents provided on PDA has so far been limited to the materials such as video and audio. MPEG-4 is the international standard used for supporting to properly save, communicate multimedia in formation such as video, audio, image, text and two-dimensional object, which can present various, multimedia contents by using adaptation techniques. However, since most MPEG-4 contents are not used for PDA but desktop, it seems not to be played on PDA where needs low power consumption, limited memory capability and GUI, and so on. In this paper, we propose the adaptation techniques, which can present the MPEG-4 contents on PDA, using scene composition with MPEG-4. The proposed scheme consists of three subparts: physical adaptation, variation adaptation and resource adaptation. Physical adaptation adjusts the physical difference of between the authoring environment and playback environment. Event adaptation part transforms events used for desktop to the events used for playback on PDA. The resource adaptation enables efficiency of playback to promote by using the essential information table on BIFS parser. As the proposed scheme is applied to MPEG-4 player, we see that the MPEG-4 contents are efficiently played on PDA.