• Proceedings of the KIEE Conference
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• 2006.10c
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• pp.428-430
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• 2006

This paper proposes a new means of performance optimization for multimedia algorithm utilizing the Microscopic DVS (Dynamic Voltage Scaling). The Microscopic DVS technique controls the operating frequency and the supply voltage levels dynamically according to the processing requirement for each frame of multimedia data. The huffman decoding algorithm of MPEG-2 AAC audio decoder is optimized to maximize the power saving efficiency of Microscopic DVS technique. The experimental results show the reduction of computational complexity by more than 30% and the reduction of power consumption by more than 17% compared with those of the conventionally fast method.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.55 no.12
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• pp.544-546
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• 2006

This paper proposes a new means of performance optimization for multimedia algorithm utilizing the Microscopic DVS (Dynamic Voltage Scaling). The Microscopic DVS technique controls the operating frequency and the supply voltage levels dynamically according to the processing requirement for each frame of multimedia data. The huffman decoding algorithm of MPEG-2 AAC audio decoder is optimized to maximize the power saving efficiency of Microscopic DVS technique. The experimental results show the reduction of computational complexity by more than 30% and the reduction of power consumption by more than 17% compared with those of the conventionally fast method.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1999.06b
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• pp.79-86
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• 1999

본 논문에서는 ISO/IEC MPEG-2 AAC 표준을 기반으로 한 멀티채널 오디오 인코더 및 디코더(MASIC) 시스템의 실시간 구현 기술에 대해서 기술한다. MPEG-2 AAC 기술은 멀티채널 오디오 부호화 방식의 국제 표준으로써, 지금까지 개발된 멀티채널 오디오 부호화 방식중 최신의 기술이며, 압축율과 오디오 품질이 가장 우수한 것으로 알려져 있다. MASIC 시스템은 인코딩 및 디코딩 기술의 실시간 처리를 위하여 범용 DSP인 TMS320C6701을 사용하였고, 멀티채널 오디오의 고속 입력과 출력을 위한 디지털 인터페이스를 가지고 있으며, 개인용 컴퓨터와의 인터페이스를 위한 PCI 기술이 적용되어 다양한 입출력 모드를 지원하는 특징을 갖는다.

• Journal of Broadcast Engineering
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• v.6 no.1
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• pp.66-71
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• 2001

This paper Introduces an implementation of MPEG-2 AAC codec system for digital audio broadcasting. This system consists of the encoder and the decoder. This system includes MPEG-2 system multiplexing and demultiplexing modules for Interfacing to the ETRI-DAB system. Four DSPs are adopted for the encoder and three DSPs for 7he decoder. Each DSP Processes system control. 1/0 control, audio signal processing. multiplexing and demultiplexing. This Paper also discusses some near future estimations relaxed to the DAB system and it\`s services. Currently a stereo audio codec is available but multi-channel audio codec and MPEG-4 audio cosec wall be also Implemented.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.255-258
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• 2001

본 논문에서는 MPEG-2 AAC(Advanced Audio Coding)와 RTP/RTCP, RTSP, TCP/IP 등의 인터넷 프로토콜을 이용한 고품질 인터넷 오디오 방송 기술에 대해 기술한다. AAC 데이터를 인터넷을 통해 실시간으로 전송하기 위해 RTP/RTCP 프로토콜을 사용하고, 사용자에게 편리한 인터페이스를 제공하기 위해 RTSP 프로토콜을 사용한다. TCP/IP 프로토콜은 서버/클라이언트간에 중요한 정보의 교환에 이용되어 진다. 본 논문에서는 위의 다양한 프로토콜을 이용하여 AAC 데이터를 스트리밍 하는 방법과 이를 이용한 인터넷 오디오 방송용 서버/클라이언트를 구성하는 방법에 대해 기술한다.

• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.693-704
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• 2011

As mobile devices become multi-functional, and converge into a single platform, there is a strong need for a codec that is able to provide consistent quality for speech and music contents. MPEG-D USAC standardization activities started at the 82nd MPEG meeting with a CfP and approved Study on DIS at the 96th MPEG meeting. MPEG-D USAC is converged technology of AMR-WB+ and HE-AAC V2. Specifically, USAC utilizes three core codecs (AAC, ACELP, and TCX) for low frequency regions, SBR for high frequency regions, the MPEG Surround for stereo information, and window transition technology for smoothing transition between various core coder. USAC can provide consistent sound quality for both speech and music contents and can be applied to various applications such as multi-media download to mobile devices, digital radio, mobile TV and audio books.

• Proceedings of the IEEK Conference
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• 2000.11c
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• pp.97-100
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• 2000

The most important technology is the compression methods in the multimedia society. Audio files are rapidly propagated through internet. MP-3(MPEG-1 Layer3) is offered to CD tone quality in 128kbps, but 64kbps below tone-quality is abruptly down. On the other hand, MPEG-II AAC (Advanced Audio Coding) is not compatible with MPEG-I, but AAC has a high compression ratio 1.4 times better than MP-3 and it has max. 7.1 channel and 96KHz sampling rate. In this paper, we designed the optimized MDCT (Modified Discrete Cosine Transform) that could decrease the capacity of enormous computation and could increase the processing speed in the MPEG-2 AAC encoder.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.251-254
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• 2001

본 논문에서는 ETRI와 연세대가 공동 개발한 AAC/MP3 복합 복호화기 ASIC 칩을 이용한 AAC/MP3 오디오 플레이어의 설계 및 구현에 대해 기술한다. 본 논문에서 사용한 AAC/MP3 복합 복호화 ASIC Chip은 20비트 고정소수점 DSP 코어를 이용하여 MP3와 MPEG-2 AAC LC 프로파일을 복호화하며, MPEG-2 AAC 메인 프로파일을 실시간으로 복호화하기 위하여 허프만 복호화 과정과 예측 과정은 전용 하드웨어 모듈을 이용하였다 이를 이용한 오디오 플레이어는 AAC/MP3 파일 재생 기능, USB를 이용한 호스트 PC와의 인터페이스 기능, Flash 메모리와의 인터페이스 기능 등의 특성을 갖는다.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.2
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• pp.33-41
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• 2001

Currently, the compression is one of the most important technology in multimedia society. Audio files arc rapidly propagated throughout internet Among them, the most famous one is MP-3(MPEC-1 Laver3) which can obtain CD tone from 128Kbps, but tone quality is abruptly down below 64Kbps. MPEC-II AAC(Advanccd Audio Coding) is not compatible with MPEG 1, but it has high compression of 1.4 times than MP 3, has max. 7.1 and 96KHz sampling rate. In this paper, we propose an algorithm that decreased the capacity of AAC encoding computation but increased the processing speed by optimizing psychoacoustic model which has enormous amount of computation in MPEG 2 AAC encoder. The optimized psychoacoustic model algorithm was implemented by C++ language. The experiment shows that the psychoacoustic model carries out FFT(Fast Fourier Transform) computation of 3048 point with 44.1 KHz sampling rate for SMR(Signal to Masking Ratio), and each entropy value is inputted to the subband filters for the control of encoder block. The proposed psychoacoustic model is operated with high speed because of optimization of unpredictable value. Also, when we transform unpredictable value into a tonality index, the speed of operation process is increased by a tonality index optimized in high frequency range.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 1999.11a
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• pp.199-203
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• 1999

This paper is on the DSP system design and implementation for real time MPEG-2 AAC multichannel audio, and MPEG-4 object oriented audio coding. This DSP system employs two DSPs of the state of the art TMS320C6701, developed by TI semiconductor. DSP board has PCI interface for downloading application program and control the system. DSP board was designed to use for both encoder and decoder, by setting several switches. The system contains external input and output box also, for A/D and D/A conversion for eight channel audio. The input box converts multi channel digital audio to ADI format, that provides serial interface for eight channel digital audio. And the output box converts ADI format signal to multi channel audio. Through this ADI interface, DSP boards can be connected to input, output box. Implemented DSP system was tested for integration with MPEG-2 AAC encoder and decoder S/W. Currently the DSP system performs realtime AAC 4-channel audio encoding with two DSPs, and 8-channel decoding with one DSP.