• Proceedings of the Korean Information Science Society Conference
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• 1998.10c
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• pp.303-305
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• 1998

MPGE(Moving Picture Expert Group)은 디지털 동영상 압축 부호화의 표준으로 자리잡고 있으며 MPEG-1에 이어 현재는 MPEG-2가 상용화되어 있는 실정이다. 복잡한 압축 기법의 적용으로 이를 재생하기 위해서는 전용의 하드웨어가 필요했지만 CPU의 성능이 향상됨에 따라 소프트웨어적으로 구현이 가능하게 되었다. 본 논문에서는 Software MPEG-1 Playback System에서 가장 큰 문제가 되는 Audio와 Video간의 동기화에 관한 기법을 제시한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3C
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• pp.240-246
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• 2008

An MPEG-2/4 AAC decoder on 16-bit fixed-point processor is presented in this paper. To meet audio quality criteria, despite small word length, special design methods for 16-bit foxed-point AAC decoder were devised. This paper presents particular algorithms for 16-bit AAC decoding. We have implemented an efficient AAC decoder using the proposed algorithms. Audio contents can be replayed in the decoder without quality degradation.

• International Journal of Fuzzy Logic and Intelligent Systems
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• v.8 no.1
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• pp.75-81
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• 2008

This paper presents a top-down approach to implement an MPEG-2 audio encoder in VLSI. As the algorithm of an MPEG-2 audio encoder is heavy-weighted and heterogeneous(to be mixture of several strategies), the encoder design process is undertaken carefully from the algorithmic level to the architectural level. Firstly, the encoding algorithm is analyzed and divided into sub-algorithms, called tasks, and the tasks are partitioned in the way of reusing the same designs. Secondly, the partitioned tasks are scheduled and synthesized to make the most efficient use of time and space. In the end, a real-time 5 channel MPEG-2 audio encoder is designed which is a heterogeneous multiprocessor system; two hardwired logic blocks and one specialized DSP processor.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.11a
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• pp.63-67
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• 1997

The ISO(International Standard Organization) has standardized MPEG-2 audio. The MPEG-2 audio compression algorithm is based upon subband analysis and exploits the human auditory characteristics to achieve a low bit rate with minimum perceptual loss of audio signal quality. This thesis presents an enhanced MPEG-2 audio encoder using multiple TMS320C30 general purpose DSP's. The developed system is made up of five slave boards and one master board. Each slave board performs susband analysis psychoacoustic parameter calculation for one channel, and the master board manages bit allocation, quantization, and bit-stream formatting for all channels. Parallel processing and pipelining techniques are used in hardware structure and fast algorithms are applied in each subroutine to implement a real-time process. The implemented system supports multichannel up to 5.1 and various bitrates.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.41 no.5
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• pp.283-290
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• 2004

In this paper, a low-power structure for 512 tap FIR filter in MPEG audio algorithm is proposed. By using CSD(Canonic Signed Digit) form filter coefficients and maximum sharing of input signal sample, it is shown that the number of adders of proposed structure can be minimized. To minimize the number of adders, the proposed structure utilizes the 4 steps of sharing, i.e., common input sharing, linear phase symmetric filter coefficient sharing, block sharing for common input, and common sub-expression sharing. Through Verilog-HDL coding, it is shown that reduction rates in the implementation area and relative power consumption of the proposed structure are 60.3% and 93.9% respectively, comparison to those of the conventional multiplier structure.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.181-190
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• 2006

Technology for compressing low-bitrate multichannel audio coding should be developed owing to the increasing need of consumer for multichannel audio contents and services. To meet this requirement, MPEG has standardized MPEG Surround. In this paper, we introduce status on MPEG Surround standardization and analyze techniques adopted in the current MPEG Surround.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• Proceedings of the Korea Multimedia Society Conference
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• 2002.11b
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• pp.771-775
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• 2002

멀티미디어 데이터의 효과적인 네트워크 전송에 대한 연구와 투자가 계속적으로 증가하고 있다. 본 논문에서는 실시간 전송 프로토콜인 RTP(Real-time Transfer Protocol)를 기반으로 MPEG 오디오 데이터를 실시간 전송하기 위한 PAD(Packet Assembler/Disassembler) 모듈을 설계, 구현하였다. RTP 기반 MPEG 오디오의 PAD 구현 방법은 MPEG 오디오의 계층에 관계없이 전송하는 방법과 MPEG Layer-3에 특화된 방법 등의 두 가지 방법이 있으며, 본 논문에서는 범용성을 중시하여 전자의 방식을 채택, 구현하였다. 구현한 PAD 모듈은 MPEG-1 및 MPEG-2 오디오 포맷을 지원한다.

• 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.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.530-533
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• 1999

During the last decade, high quality digital audio has essentially replaced analog audio. During this period, digital audio have applied many application areas of the info-industry. These applications have created a demand for high quality digital audio. In audio compression, the methods using human auditory nervous properties are used and introduced from psychoacoustical model utilized perceptual audio coding unable to code above the limitation of human perception. The discussion concentrates on architectures and applications of those techniques which utilize psychoacoustical models to exploit efficiently masking characteristics of the human receiver. In this paper, the designed MDCT/IMBCT as a standard of current MPEG is implemented onto FPGA.