• 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 the Institute of Convergence Signal Processing
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• v.2 no.1
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• pp.113-118
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• 2001

In this paper, we implement modules of the MPEG-2 AAC decoder using VHDL. Tools of Huffman decoder, inverse quantizer and high-density filter bank which are necessary for the AAC decoder. We designed the high speed Huffman decoder using the method of octal tree search algorithm, and reduced computational time of filter bank using IFFT. Also, we use table of computation result for an exponential calculation of Inverse quantizer in fixed-point hardware, and reduced the size of table using linear interpolation. Modules implemented by hardware through optimization work in real time at low clock frequency are possible to reduce the system size.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.178-181
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• 2000

In this paper, we present the implementation of filterbank for MPEG-2 Advanced Audio Coding (AAC) decoder with VHDL. The filterbank of AAC employs a technique called time-domain aliasing cancellation (TDAC). In order to make the algorithm more efficiently, we decompose and reorganize the filterbank algorithm lot the high speed decoding process and lower computational cost. And we make this filterbank algorithm to be used with other modules of AAC decoder in parallel processing.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.874-882
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• 2004

MPEG Audio compression algorithm is the international standard for the digital compression of high quality audio using mechanism of the perceptual coding based on psychoacoustic masking. It is necessary to discuss the constraints on designing of common filter banks for MPEG-2 BC and MPEG-2 AAC decoder system, which is not Down yet, mapping audio signals from the time domain into the frequency domain. In this paper, we present an architecture of common synthesis filter whcih can be used for MPEG-2 BC and MPEG-2 AAC decoder using recursive structure. The proposed algorithm is based on recursive architecture that effectively performs common compulsion.

• 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 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.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.257-260
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• 2001

본 논문에서는 2 채널 MAIN 프로필 MPEG-2 AAC 복호화 시스템의 구조를 제안하고 구현하였다. 복호화 알고리듬의 구조적인 모듈화에 근거하여, 시스템 설계 과정에서 전체 시스템을 3 개의 하드웨어 모듈로 분할하였다. 전체 시스템은 허프만 복호화기, 예측기, 20 비트 고정소수점 DSP 코어로 이루어져 있다. 허프만 복호화기는 주어진 작업을 1 클럭 사이클 내에 수행할 수 있는 고속의 하드와이어드 모듈이고, 예측기는 높은 해상도를 가지고 다른 모듈들과 병렬처리가 가능한 구조를 가진 모듈이다. 구현된 시스템은 16.9 MIPS 로 2 채널의 MPEG-2 AAC 비트열을 고음질로 복호화할 수 있다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.247-250
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• 2000

본 논문에서는 MPEG-2 AAC 오디오 복호화 시스템을 효율적으로 설계하고 구현하였다. 구현된 시스템은 2채널의 메인 프로필 MPEG-2 AAC 비트열을 실시간으로 복호화하고, 32, 44.1, 48kHz의 표본화 주파수를 지원하여, 표준안에서 제안하는 툴 중 커플링 채널을 제외한 모든 툴을 지원한다. 설계된 시스템은 허프만 복호화와 예측 과정을 수행하는 두개의 독립된 모듈과 Programmable DSP 코어의 혼합 구조(hybrid architecture)로 최적화된 구조를 갖는다.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.1099-1102
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• 1999

This paper presents a real-time MPEG-2 AAC decoding system, which can decode 2-channel main profile MPEG-2 AAC bitstream. The proposed system supports all decoding tools except for coupling channel tool, and provides sampling rates of 32, 44.1, 48 KHz. The system consists of a simple programmable DSP core and two hardwired logic modules that perform Huffman decoding and prediction for real-time implementation.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.08a
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• pp.173-176
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• 2000

본 논문은 VHDL을 이용하여 1997년 국제 표준안으로 제정된 MPEG-2 AAC 복호화기의 각 모듈을 구현하였다. AAC 복호화기는 허프만 복호화, 역양자화, 고해상도 필터뱅크 등의 도구들이 필수적으로 사용된다. AAC 복호화기의 실시간 구현을 위해 각 도구들의 알고리즘을 분석하고, 하드웨어 개발에 알맞게 최적화하여 고속화와 적은 메모리를 사용하여 효율적으로 구현하였다.