• 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.

• Journal of Korea Society of Industrial Information Systems
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• v.12 no.1
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• pp.54-60
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• 2007

The Digital Multimedia Broadcasting(DMB) system is developed to offer high quality multimedia content to the mobile environment. The system adopts the MPEG-4 standard for the main video, audio and other media format. For providing interactive contents, it also adopts the MPEG-4 scene description that refers to the spatio-temporal specifications and behaviors of individual objects. With more interactive contents, the scene description also needs higher bitrate. However, the bandwidth for allocating meta data, such as scene description is restrictive in the mobile environment. On one hand, the DMB terminal renders each media stream according to the scene description. Thus the binary format for scene(BIFS) stream corresponding to the scene description should be decoded and parsed in advance when presenting media data. With this reasoning, the transmission delay of the BIFS stream would cause the delay in transmitting whole audio-visual scene presentations, although the audio or video streams are encoded in very low bitrate. This paper presents the effective optimization technique in adapting the BIFS stream into the expected bitrate without any waste in bandwidth and avoiding transmission delays inthe initial scene description for interactive DMB content.

• The KIPS Transactions:PartB
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• v.17B no.2
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• pp.99-106
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• 2010

In this paper, we propose the construction of multi-queries and consistency principle for the user's emotion-based music retrieval system. The features used in the system are MPEG-7 audio descriptors, which are international standards recommended for content-based audio retrievals. In addition we propose the method to determine the weight that represent the importance of each descriptor for each emotion in order to reduce the computation. Also, the proposed retrieval algorithm that uses the relevance feedback based on consistency principal and multi-queries improves the success ratio of musics corresponding to user's emotion.

• The Magazine of the IEIE
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• v.28 no.4
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• pp.103-104
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• 2001

• Proceedings of the IEEK Conference
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• 2002.07a
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• pp.125-128
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• 2002

In this paper, a new interactive TV service model is proposed. The MPEG-4 system is specified for composing and managing various object streams including user interactions. The data broadcasting model supporting user interactions is designed using MPEG-4 system in our proposal. We evaluate possibility of proposed service model using simulation player. This player supports MPEG-2 TS which contains MPEG-2 video and AC-3 audio streams as a main service and MPEC-4 system data as interactive services as well as user specific EPG information, and XML data, etc as supplemetary services. The player also supports a multi-channel environment. The synchronization between audio and visual data is achieved by DTS and PTS in TS.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9C
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• pp.851-859
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• 2003

최근 사용자들의 멀티채널 선호도는 급속도로 전파되고 있다. MPEG은 동영상 및 음향시스템의 데이터 압축기술을 제공하는데, 현재 각광을 받고있는 것이 디지털 오디오이다. MPEG 표준안은 MPEG-1오디오 알고리즘을 MPEG-2 알고리즘에 동일하게 사용해도 멀티채널 및 5.1채널 사운드륵 제공한다. MDCT(Modified Discrete Cosine Transform)는 TDAC(Time Domain Aliasing Cancellation)에 기반을 두고있는 변형이산 여현 변환을 나타낸 것이다. 본 논문에서는 오디오 부분의 핵심이라 할 수 있는 MDCT/IMDCT(Inverse MDCT) 알고리즘을 최적화하여 효율적인 알고리즘을 제안하였다. 그리고 연산과정에서 중복되는 영역을 묶음으로써 연산에 필요한 계수를 줄였다. 최적화 전에 비해 코사인 계수를 0.5%이하로 최적화하였고, 승산에서 0.098%, 가산에서 0.58% 효율을 보였다. 알고리즘 검증은 C언어를 사용하여 검증하였고, 최적화된 알고리즘을 적용하여 마이크로 프로그램 방식의 하드웨어 구조론 설계하였다.

• Proceedings of the IEEK Conference
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• 2000.11d
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• pp.119-122
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• 2000

Recently, many complex DSP (Digital Signal Processing) algorithms have being realized on RISC CPU due to good compilation, low power consumption and large memory space. But, real-time implementation of multiple DSP algorithms on RISC requires the minimum and efficient memory usage and the lower occupancy of CPU. In this thesis, the original floating-point code of MPEG-1 audio decoder is converted to the fixed-point code and then optimized to the efficient assembly code in time-consuming function in accord with RISC feature. Finally, compared with floating-point and fixed-point, about 30 and 3 times speed enhancements are achieved respectively. And 3~4 times memory spaces are spared.

• Journal of Broadcast Engineering
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• v.13 no.4
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• pp.452-462
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• 2008

In this paper, we address issues associated with the real-time implementation of the MPEG-1/2 Layer-II (or MUSICAM) and MPEG-4 ER-BSAC decoders for Digital Multimedia Broadcasting (DMB) on TMS320C64x+ that is a fixed-point DSP processor with a clock speed of 330 MHz. To achieve the real-time requirement, they should be optimized in different steps as follows. First of all, a C-code level optimization is performed by sharing the memory, adjusting data types, and unrolling loops. Next, an algorithm level optimization is carried out such as the reconfiguration of bitstream reading, the modification of synthesis filtering, and the rearrangement of the window coefficients for synthesis filtering. In addition, the C-code of a synthesis filtering module of the MPEG-1/2 Layer-II decoder is rewritten by using the linear assembly programming technique. This is because the synthesis filtering module requires the most processing time among all processing modules of the decoder. In order to show how the real-time implementation works, we obtain the percentage of the processing time for decoding and calculate a RMS value between the decoded audio signals by the reference MPEG decoder and its DSP version implemented in this paper. As a result, it is shown that the percentages of the processing time for the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders occupy less than 3% and 11% of the DSP clock cycles, respectively, and the RMS values of the MPEG-1/2 Layer-II and MPEG-4 ER-BSAC decoders implemented in this paper all satisfy the criterion of -77.01 dB which is defined by the MPEG standards.

• Journal of Broadcast Engineering
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• v.11 no.2 s.31
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• pp.191-199
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• 2006

This paper presents an efficient pre echo control scheme for MPEG Audio coders based on the psychoacoustic model II (PAM-II). Pre echo control is the final step for the calculation of masking threshold in the PAM II. It is to minimize the spread of quantization error over the processing frame. In the conventional encoders, pre echo is reduced by restricting the estimated masking threshold not to exceed the one obtained in the previous frame. The conventional method performs pre echo control not only for short blocks but also for long blocks, which lowers the masking threshold in long blocks and, in turn, increases the quantization noise level of corresponding blocks. This paper proposes an efficient pre echo control process. The test result shows a mean enhancement of more than 0.4 especially for complex signals on the ITU R 5 point audio impairment scale.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.5
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• pp.112-117
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• 2007

This paper presents specialized DSP instructions and their hardware architecture for audio coding algorithms, such as the MPEG-2/4 Advanced Audio Coding(AAC), Dolby AC-3, MPEG-2 Backward Compatible(BC), etc. The proposed architecture is specially designed and optimized for the MDCT/IMDCT(Inverse Modified Discrete Cosine Transform), and Huffman decoding of the AAC decoding algorithm. Performance comparisons show a significant improvement compared with TMS320C62x and ASDSP21060 for the MDCT/IMDCT computation. In addition, the dedicated Huffman decoding accelerator performs decoding and preparing operand in only one cycle. The proposed DPU(Data Processing Unit) consists of 107,860 gates and achieves 150 MIPS.