• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.659-662
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• 1999

In this paper we implement MP3 encoder based on integer operations. To implement MP3 encoder presented in ［1］, floating-point operations are required. But we devise an MP3 encoding method which is based on integer operations. To verify the method presented in this paper, we implement MP3 encoder using ARM processor. In this paper we present the method to change floating point operations into integer operations, and the ARM assembly programming technique to implement fast MP3 encoder. The MP3 encoder implement using integer processor consumes less power than the encoder implemented using floating-point processor. So the encoder implemented in this paper is suitable lot portable applications which requires low power consumption.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.237-240
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• 2001

MP3는 현재 디지털 오디오 압축 표준으로 널리 사용되고 있으며 12:1의 높은 압축률을 가진다. MP3 encoding의 경우 현재는 대부분 PC를 이용하고 있으므로 MP3 encoding과 decoding을 동시에 할 수 있는 칩이 나오면 이러한 불편함 없이 portable이 가능해지며 라디오, CD, 카세트 테이프 등으로부터 고음질 녹음 후 재생이 가능해진다. TMS320C30과 호환되는 DSP 코어를 이용하였으며 MP3 엔코딩과 디코딩 실시간 수행을 위하여 알고리즘 개선과 FFT block, 주변 interface block을 설계하였다. 최종적으로 MP3 encoder와 decoder 칩 설계하고 이를 하드웨어 에뮬레이션을 이용하여 검증하였다.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.10
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• pp.199-203
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• 2013

In this paper, a fast algorithm of discrete cosine transform-based interpolation filter (DCT-IF) for HEVC (high efficiency video coding) encoder is proposed. DCT-IF filter accounts for around 30% of encoder complexity, according to the computational complexity analysis with the HEVC reference software. In this work, the proposed DCT-IF is optimized by applying frame-level interpolation, SIMD optimization, and task-level parallelization via OpenMP on a developed C-based HEVC encoder. Performance analysis is conducted by measuring speed-up factor of the proposed optimization technique on the developed encoder. The results show that speed-up factors by frame-level interpolation, SIMD, and OpenMP are approximately 38-46, 3.6-4.4, and 3.0-3.7, respectively. In the end, we achieved the speed-up factor of 498.4 with the proposed fast algorithm.

• The Journal of the Acoustical Society of Korea
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• v.25 no.2
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• pp.61-71
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• 2006

An MP3 encoder chip has been designed and fabricated with the hardware and software co-design concepts. In the aspect of the software. the calculation cycles of the distortion control loop. which requires most of the calculation cycles in MP3 encoding procedure. have been reduced to $67\%$ of the original algorithm through the 'scale factor Pre-calculation'. By using a floating Point 32 bit DSP core and designing the FFT block with the hardware. we can get the additional reduction of the calculation cycles in addition to the software optimization. The designed chip has been verified using HW emulation and fabricated via 0.25um CMOS technology The fabricated chip has the size of $6.2{\time}6.2mm^2$ and operates normally on the test board in the qualitative and quantitative aspect.

• 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.

• 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.

• Journal of Korea Multimedia Society
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• v.8 no.7
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• pp.882-888
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• 2005

MPEG-2 Advanced Audio Coding (AAC) has already been standardized as a sophisticated next generation technology AAC provides an audio signal that has CD quality at 96-128kbps/stereo. This paper describes a high-quality and efficient software implementation of an MPEG-2 AAC LC Profile encoder. Common scalefactor and noisless coding are accelerated by $45\%$ and $27\%$, respectively, through the use of TMS320C30 instructions. The implemented encoder uses 7.5kWords of program memory, 18kWords of data ROM and 92kBytes of data RAM, respectively. The results of subjective Qualify test showed that the sound quality achieved at 96kbps/stereo was equivalent to that of MP3 at 128kbps/stereo.

• 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 IEEK Conference
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• 2000.06c
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• pp.93-96
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• 2000

Currently, the most important technology is the compression methods in the multimedia society. Audio files are rapidly propagated through internet. MP-3 is offered to CD tone quality in 128Kbps, but 64Kbps below tone quality is abruptly down and high bitrate. 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 better than MP-3. Especially, AAC has max. 7.1 channel and 96KHz sampling rate. In this paper, the perceptual model is dealt with 44.1KHz sampling rate for SMR(Signal to Masking Ratio)

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.469-472
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• 2001

MPEC Audio Layer-III(MP3)알고리듬은 복호화기에 비해 부호화기가 월등히 많은 연산량을 가지고 있는 비대칭 구조를 가지고 있다. MP3 부호화기의 대부분의 연산량은 복잡한 초월함수 연산이 포함되는 심리음향모델과 반복 루프 과정을 수행하는 비선형 양자화와 비트 할당과정 이 차지한다. 본 논문에서는 MP3 부호화기의 실시간 구현을 위한 알고리듬 레벨의 최적화를 수행하였다. MP3 부호화기의 연산량을 줄이기 위해 심리음향모델을 간략화하고 반복 루프의 회수를 최소화할 수 있는 방법을 제안하였다. 프레임당 한 그래뉼의 심리음향모델 정보를 계산하여 한 프레임 내에서의 심리음향모델 정보를 추정함으로써 연산량을 45% 이상 감소시켰다. 또한 외부 반복 루프의 반복 회수를 줄이기 위하여 외부 반복 루프의 반복에 따른 스케일 팩터(Scale Factor) 및 양자화 스탭의 증가 패턴을 관찰하고 최적화된 스캐일 팩터 증가 방법을 제안하였다. 제안된 고속화 방법은 주관적 음질 평가를 통해 성능을 검증하였다.