• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.865-873
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• 2004

MPEG-2 Advanced Audio Coding(AAC) is widely used in the multi-channel audio compression standards. And it combines hi인-resolution filter bank prediction techniques, and Huffman coding algorithm to achieve the broadcast-quality audio level at very low data rates. The forward and inverse modified discrete transforms which are operated in the encoder and the decoder of the filter bank need many computations. In this paper, we propose suitable recursive structure at IMDCT processing for MPEG-2 AAC real-time decoder. We confirm the memory, the computation speed and complexity of the proposed structure.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.149-153
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• 1995

본 논문에서는 TI(Texas Instrument)사의 범용 디지탈 프로세서인 TMS320C30을 이용하여 MPEG-2 계층2(Layer II) 오디오 부호화 알고리듬의 실시간 처리가 가능한 시스템을 구현하였다. 구현한 시스템은 1 채널의 오디오 신호를 처리하기 위한 Slave 보드 5개와 채널 멀티플렉싱과 부가 처리를 위한 Master 보드 1개로 이루어져 있다. MPEG-2 알고리듬의 각 단계별 소요시간을 계산한 후, 이를 바탕으로 각 프로세서에 할당하는 작업량을 조정하여 실시간 처리에 적합한 시스템을 구현하였다.

• Journal of Broadcast Engineering
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• v.28 no.2
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• pp.213-229
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• 2023

In this paper, we introduce a spatial sound rendering system that provides 6DoF spatial sound in real time in response to the movement of a listener located in a virtual environment. This system was implemented using MPEG-I AEP as a development environment for the CfP response of MPEG-I Immersive Audio and consists of an encoder and a renderer including a decoder. The encoder serves to offline encode metadata such as the spatial audio parameters of the virtual space scene included in EIF and the directivity information of the sound source provided in the SOFA file and deliver them to the bitstream. The renderer receives the transmitted bitstream and performs 6DoF spatial sound rendering in real time according to the position of the listener. The main spatial sound processing technologies applied to the rendering system include sound source effect and obstacle effect, and other ones for the system processing include Doppler effect, sound field effect and etc. The results of self-subjective evaluation of the developed system are introduced.

• 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 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.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.42 no.4 s.334
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• pp.61-68
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• 2005

A chip, which can fast encoder the audio data to AAC (Advanced Audio Coding) LC(Low Complexity) that is MPEG-2 audio standard, has been designed on the basis of a 32 bits DSP core and fabricated with 0.25um CMOS technology. At first, the various optimization methods for implementing the algerian are devised to reduce the memory size and calculation cycles. FFT(Fast Fourier Transform) hardware block is added to the DSP core to get the more reduction of the calculation cycles. The chips has the size of $7.20\times7.20 mm^2$ and about 830,000 equivalent gates, can carry out AAC encoding under 70MIPS(Million Instructions per Second).

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.4 s.304
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• pp.167-176
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• 2005

This paper proposes a new power minimization technique for the frame-based multimedia signal processing. The derivation of the technique is based on the newly proposed microscopic DVS(Dynamic Voltage Scaling) method, where, the operating frequency and the supply voltage levels are dynamically controlled according to the processing requirement for each frame of multimedia data. The multimedia signal processing algorithms are also redesigned and optimized to maximize the power saving efficiency of the microscopic DVS technology. The characterization of the mean/variance distribution of the processing load in the frame-based multimedia signal processing provides the major basis not only for the optimized application of the microscopic DVS technology but also for the optimization of the multimedia algorithms. The power saying efficiency of the proposed DVS approach is experimentally tested with the algorithms of MPEG-2 video decoder and MPEG-2 AAC audio encoder on the ARM9 RISC processor. The experimental results with the diverse MPEG-2 video and audio files show The average power saving efficiencies of 50$\%$ and 30$\%$, respectively. The results also agree very well with those of the analytic derivations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.11C
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• pp.1491-1497
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• 2004

The psychoacoustic model in MPEG audio layer-III (MP3) encoder is optimized for the fixed-point processing. The optimization process consists of determining the data word length of arithmetic unit and the algorithm for transcendental functions that are often used in the psychoacoustic model. In order to determine the data word length, we defined a statistical model expressing the relation between the fixed-point operation errors of the psychoacoustic model and the probability of alteration of the allocated bits doe to these errors. Based on the simulations using this model, we chose a 24-bit data path and constructed a 24-bit fixed-point MP3 encoder. Sound quality tests using the constructed fixed-point encoder showed a mean degradation of -0.2 on ITU-R 5-point audio impairment scale.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.205-208
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• 1998

본 논문에서는 VLSI 기술에 바탕을 둔 top-down 접근 방식에 의하여 MPEG-2 오디오 부호화 알고리듬을 구현하였다. MPEG-2 오디오 부호화기의 알고리듬은 많은 연산량을 갖고 이질적인 특성을 갖고 이질적인 특성을 갖는 알고리듬들이 복합적으로 존재한다. 그러므로, 부호화기를 효과적으로 구현하기 위해서는 알고리듬 수준에서 구조적 수준에 이르기까지 많은 고찰이 이루어져야 한다. 본 논문에서는 우선 전체 부호화 알고리듬을 분석하여 이들을 다시 작업이라고 정의된 작은 부-알고리듬으로 나누었다. 다음으로, 분할된 작업들은 시간과 공간을 초대한 활용할 수 있도록 적절한 작업 순서를 부여하고, 좀 더 큰 모듈들로 모으는 클러스터링을 수행하였다. 마지막으로 이러한 분석 결과를 바탕으로, 실시간으로 동작하는 5.1 채널 MPEG-2 오디오 부호화기를 설계하였다. 설계된 시스템은 두 개의 하드웨어 블록과 한 개의 ASIP형 DSP 프로세서를 갖는 이질적인 다중 프로세서의 형태를 갖는다. 설계된 오디오 부호화기는 0.6$\mu\textrm{m}$ 표준 셀 기술을 이용하여 단일 칩으로 제작되었으며, PC에 탑재 가능한 시험 기판을 제작하여 동작을 검증하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.217-222
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• 1999

본 논문에서는 멀티채널 오디오 부호화 방식인 MPEG-2 AAC(Advanced Audio Coding) 국제 표준을 수용한 AAC 인코더 및 디코더의 실시간 구현에 대해 기술한다. 범용 DSP 인 TMS320C6701 DSP를 이용한 하드웨어 플랫폼과 이 플랫폼에서 실시간으로 동작되는 인코더와 디코더 소프트웨어를 설계, 개발(MASIC 시스템)하였다. 구현한 MASIC 시스템은 오디오 입력 장치, 출력 장치, 인코더 보드, 그리고 디코더 보드로 구성되어 있으며, 개인용 컴퓨터의 PCI 슬롯을 이용하여 인코더의 경우 최대 6채널의 오디오를, 디코더의 경우 8채널의 오디오를 실시간 동작으로 처리할 수 있다. 인코더 및 디코더의 실시간 처리를 위한 소프트웨어 최적화 기술 및 인코더와 디코더의 연동시험에 대해서도 기술하며, 개인용 컴퓨터에서 실시간으로 수행되는 스테레오 AAC 디코더 소프트웨어의 개발 결과를 기술한다.