• 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 Electronics and Information Engineers
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• v.50 no.1
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• pp.86-96
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• 2013

Unified Speech and Audio Coding (USAC) is the speech/audio codec with the best quality, approved on Final Draft International Standard (FDIS) at MPEG meeting in 2011. Since MPEG conventionally standardizes only the decoder, it is not easy to study on the encoder technologies. Furthermore, Reference Model(RM) shows extremely poor performance. To solve these problems, the open source project(JAME) proposes the methods to make the improved performance of main encoder technologies in USAC. Especially, this paper introduces the encoder modules: the signal classifier for selective operation between two coders, the psychoacoustic model in frequency domain, and window transition technology. Finally, the results of verification test for FDIS and the performance of Common Encoder are appended.

• 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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• 1999.06b
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• pp.137-142
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• 1999

지금까지 알려진 오디오 부호화 방식중 실제로 스케일러블 부호화 방식은 없었다. MPEG-4에서는 스케일러블 부호화라는 새로운 기능을 중요한 요구사항으로 받아들여 개발하였다. 스케일러블 부호화(scalable coding) 기능이란 주어진 자원의 일부분을 가지고 의미 있는 정보를 재생할 수 있게 하는 부호화방식이다. 본 논문에서는 ISO MPEG-4에서 추진하고 있는 스케일러블 부호화방식인 large step scalable coding방식과 fine grain scalable coding방식에 대해 알아보고 그 응용분야에 대해 살펴본다.

• 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.

• 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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• 2006.06a
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• pp.241-242
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• 2006

MPEG Surround is an audio coding technology that represents multi-channel audio signal with downmixed audio signal(s) and very low bitrate side information based on Binaural Cue Coding. The side information consists of Inter-Channel Level Difference, Inter-Channel Correlation, and payloads. These two parameters are correspondent to the well-known spatial parameters in psycho-acoustics, Inter-aural Level Difference (ILD) and Inter-Aural Cross Correlation (IACC). Though ICLD is to provide perceptually equivalent ILD to the listener, however, the ILD of the original multi-channel audio signal and that of the MPEG Surround encoded signal was different. The difference between two ILD values is defined as ILD Distortion (ILDD). This paper provides how ILDD can be applied to enhance sound quality in MPEG Surround and how much ILDD is decreased.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06c
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• pp.249-254
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• 1998

MPEG 오디오 그룹에서는 오디오 부호화 기술의 국제 표준으로 MPEG-1 오디오, MPEG-2 오디오 BC, MPEG-2 AAC의 규격 제정을 완료하였고, 현재 MPEG-4 오디오 및 MPEG-7 오디오의 국제 표준을 제정하고 있다. 본 논문에서는 이들 표준에 대한 요구 기능 및 기술 특징을 분석하고, 각각의 표준에 대한 응용분야와 향후의 계획에 대하여 기술한다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.1 no.2
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• pp.91-98
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• 2008

Domestic standard for Korean T-DMB includes MPEG-4 BSAC (Bit Sliced Arithmetic Coding) audio coding that has been established in 2003. This paper presents an implementation and optimization of MPEG-4 BSAC Audio Decoder on ARM926EJ-S processor. Tools and modules of the BSAC audio decoder were implemented with 32-bit fixed point operations. Further optimization was accomplished using ARM926EJ-S Inline Assembly. The optimization was based on the total number of multiplications and MAC (Multiply and Accumulation) operations causing most of core cycles of ARM926EJ-S, and also based on analysis of ARMv5 instructions. The result of optimization was evaluated on the basis of MIPS (Million Instruction per second). Implementation results show that BSAC bitstream at 96kbps can be decoded in real-time at 65MHz CPU clocks.

• Proceedings of the Korea Multimedia Society Conference
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• 2004.05a
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• pp.370-373
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• 2004

The rapid proliferation of mobile device such as PDA allows users more ubiquitous access to multimedia information. The user mobility provides users a uniform vision of their preferred working environment independently of their current points of attachment. Supporting the user mobility requires the Player capable of efficiently presenting the multimedia contents. MPEC-4 provides not only the description for coding audio and video (as its predecessors MPEG-1 and MPEG-2), but also for coding images, animations, interactivity and protecting content. With MPEG-4, we present interactive media using multiple objects - audio, video, image, 2D geometry, and text - in a single format. Therefore we propose the MPEC-4 Player for PDA. The proposed MPEG-4 Player for PDA supports mobility, portability and personality.