• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.538-541
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• 1999

MPEG-2 AAC(Advanced Audio Coding) is the most advanced coding scheme available for high quality audio coding. This MPEG-2 AAC audio Standard allows for ITU-R ‘indistinguishable’ quality according to at data rates of 320 kb/s for five full-bandwidth channel audio signals. The compression ratio is around a factor of 1.4 better compared to MPEG Layer 3, you get the same quality at 70% of the bitrate. This paper suggest optimization method for MDCT/IMDCT (Modified Discrete Cosine Transform/Inverse Modified Discrete Cosine Transform) in Encoder and Decoder for AAC.

• 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.

• Journal of Korea Society of Industrial Information Systems
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• v.13 no.5
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• pp.117-123
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• 2008

In this paper, we develop MPEG-4 AAC streaming player on embedded Linux processor such as mobile terminals. Moreover we show the experimental results that the player preforms the decoding processes of MPEG-4 AAC data effectively. MPEG-4 AAC technology supports a wide range encoding rates and high sound quality so it is appropriate to adopt various applications. In particular, the need in the development of the application of audio data increases according to significantly increase in devices used in mobile environments, such as cell phones and PDAs. In this environment, it is necessary to optimize the decoding processes to the ability of the terminal hardware in order to play audio data without delays. We also implement the decoding module to optimize the processor capabilities and make the player to decode and play streaming audio data from streaming server.

• Proceedings of the KIEE Conference
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• 2006.07d
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• pp.2047-2048
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• 2006

본 논문에서는 MPEG-4 AAC 표준에 근거하여 개인휴대단말과 같은 모바일 단말의 응용을 위해 연산 복작도를 낮춘 새로운 양자화기 구조를 제안하였다. 기존 MPEG-4 AAC 인코더의 양자화기는 rate control과 distortion control을 동시에 수행하는 순환 이중루프 구조를 갖으며, 높은 연산부담으로 인해 실시간 인코딩 및 시스템의 소비전력을 증가시키는 중요한 요인으로 작용한다. 본 논문에서는 이러한 제약조건을 극복하기 위해 허용 잡음 레벨과 대역별 양자화 잡음을 정합하는 잡음 포락선 정합기법을 제안하였고, 이를 응용하여 낮은 소비전력 실현이 가능한 단일루프 구조의 단순화된 양자화기 구조를 제안하였다. 또한 제안한 양자화기를 적용한 독자적인 구조의 MPEG-4 AAC 인코더를 구현하여 검증 및 성능평가를 수행하였다.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.329-332
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• 1999

Currently, the most important technology is compression methods in the multimedia society. In audio compression, the method using human auditory nervous property is used. This method using psychoacoustical model is applied to perceptual audio coding, because human's audibility is limited. MPEG-II AAC(Advanced Audio Coding) is the most advanced coding scheme that is of benefit to high quality audio coding. The compression ratio is 1.4 times compared with MPEG-I layer-III. In this paper, the vector processor for MDCT/IMDCT(Modified Discrete Cosine Transform /Inverse Modified Discrete Cosine Transform) of MPEG-II AAC is designed.

• The Journal of the Acoustical Society of Korea
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• v.21 no.3
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• pp.237-244
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• 2002

In this paper we mention the optimization of MPEG-4 VM (Moving Picture Expert Group-4 Verification Model) GA (General Audio) AAC (Advanced Audio Coding) encoder and the design of the decoder for PDA (Personal Digital Assistant) using MPEG-4 VM source. We profiled the VMC source and several optimization methods have applied to those selected functions from the profiling. Intel Pentium III 600 MHz PC, which uses windows 98 as OS, takes about 20 times of encoding time compared to input sample running time, with additional options, and about 10 times without any option. Decoding time on PDA was over 35 seconds for the 17 seconds input sample. After optimization, the encoding time has reduced to 50% and the real time decoding has achieved on PDA.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.39 no.4
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• pp.355-355
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• 2002

In this paper, we designed a FPU(floating point unit) that it is very important and requires of high density when digital audio is designed. Almost audio system must support the multi-channel and required for high quality. A floating point arithmetic function in MPEG-2 AAC that implemented by hardware is able to realtime decoding when DSP realization. The reason is that MPEG-2 AAC is compatible to the Audio field of MPEG-4 and afterwards. We designed a FPU by hardware to increase the speed of a floating point unit with much calculation part in the MPEG-2 AAC Decoder. A FPU is composed of a multiplier and an adder. A multiplier used the Radix-4 Booth algorithm and an adder adopted 1's complement method for speed up. A form of a floating point unit has 8bit of exponent part and 24bit of mantissa. It's compatible with the IEEE single precision format and adopted a pipeline architecture to increase the speed of a processor. All of sub blocks are based on ISO/IEC 13818-7 standard. The algorithm is tested by C language and the design does by use of VHDL(VHSIC Hardware Description Language). The maximum operation speed is 23.2MHz and the stable operation speed is 19MHz.

• Journal of Broadcast Engineering
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• v.13 no.3
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• pp.415-418
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• 2008

Some embedded microprocessors do not have an FPU(Floating Point Unit) due to a circuit complexity and power consumption. The performance speed of MPEG-4 AAC decoder on this hardware environment would be slower than corresponding speed for playing back of the decoded results. Therefore, irritating and high-pitched noises are interleaved in the original the audio data. So, in order to play MPEG-4 AAC file on such PDA, a new algorithm that transforms floating-point arithmetic to one with integers, is needed. We have developed a transformation algorithm from floating-point operation to integer operation and implemented the PDA's AAC Player. We also show the efficiency of our proposed method with the experimental results.

• 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)

• Journal of Electrical Engineering and Technology
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• v.9 no.4
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• pp.1411-1417
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• 2014

An MPEG-2 AAC Huffman decoding method based on the fixed length compacted codeword tables, where each codeword can contain multiple number of Huffman codes, was proposed. The proposed method enhances the searching efficiency by finding multiple symbols in a single search, i.e., a direct memory reading of the compacted codeword table. The memory usage is significantly saved by separately handling the Huffman codes that exceed the length of the compacted codewords. The trade-off relation between the computational complexity and the amount of memory usage was analytically derived to find the proper codeword length of the compacted codewords for the design of MPEG-2 AAC decoder. To validate the proposed algorithm, its performance was experimentally evaluated with an implemented MPEG-2 AAC decoder. The results showed that the computational complexity of the proposed method is reduced to 54% of that of the most up-to-date method.