• The KIPS Transactions:PartC
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• v.11C no.6 s.95
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• pp.773-780
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• 2004

In this paper, we measured the transmission characteristics of a MPEG2 and a DV that are typical stream-type traffics on the very high speed network and carried out the subjective evaluation of end users for these stream-types. In the subjective evaluation of these stream-type data, video quality evaluation is based on ITU-R BT.500-1 and audio qualify evaluation is based on ITU-R BS.1116-1. Also experiment method to acquire the subjective evaluation of end users is selected the 5 grades method of DSCQS. Under the same condition, in case of MPEG2, the evaluation grade of the video and the audio duality becomes deteriorated at the load rate of $54\%$ that network traffic increases rapidly. In case of DV the evaluation grade of video duality began decrease, but the degree of the change was slower than MPEG2 at the same load rate. Moreover the subjective evaluation grade of end users was superior to load rate $70\%$ in case of DV audio quality, traffic and QoS control that consider the subjective evaluation of end user is required. Conclusively, in case of MPEG2, we can perform traffic control that only use the actual measurement values on the network. However in case of DV, we can perform traffic control that the actual measurement values on the network and the subjective evaluation of end users are considered at the same time.

• Journal of Broadcast Engineering
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• v.21 no.3
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• pp.369-379
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• 2016

This paper deals with an efficient method for skipping inter-channel phase differences (IPD) in the MPEG surround of the unified speech and audio coding (USAC). Based on the psycho-acoustic sensitivity on the IPD, we estimate a threshold on IPD, below which we can not notice degradation in spatial cue. We propose an IPD skip method, in which any IPDs within the threshold are set to zero and are not transmitted. The proposed IPD skip method gives about 38% savings in terms of bit amount for IPD. Nevertheless, in the MUSHRA test, the proposed method does not show any noticeable degradation in the decoded audio quality.

• The Journal of Engineering Research
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• v.2 no.1
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• pp.57-62
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• 1997

Two psychoacoustic models used in MPEG-1 employ different masking patterns, different masking indexes, and different computational procedures. As a result, Model 1 is inferior to Model 2 due to its worst case approach in computing the SMR even though it determines tonality and masking levels accurately. In this study, we investigate the performances of psychoacoustic models when we modify the MPEG-1 audio coder for variable bitrates. Simulation results show that Model 2 has a gain of 30 kbps in the dual channel mode and 20 kbps in the joint stereo mode. It is generally known that the joint stereo mode has a gain in bitrate compare to the dual channel mode. For signals with frequent attacks, this gain becomes larger in Model 1 than in Model 2. This is due to the fact that Model 1 uses the worst case approach in computing the SMR to reduce pre-echo

• Proceedings of the IEEK Conference
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• 2003.11b
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• pp.251-254
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• 2003

DMB is a next-generation multimedia broadcasting system that not only enables digital broadcasting services such as transmission of CD-duality audio, traffic information, and real-time stock information, but also allows reception of high-quality digital TV in high-speed driving conditions. In the DMB system, MPEG-2 TS(Transport Stream) multiplex method and MPEG-4 System SL(Sync Layer) have been selected as the delivery layer. In this paper, an efficient interface scheme between an MPEG-2 TS processing hardware and software-implemented MPEG-4 system within a DMB terminal device is proposed.

• 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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• 1999.11a
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• pp.1123-1126
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• 1999

In this paper, we propose a real-time implementation of the MPEG-1 layer III and MPEG-2 layer III LSF audio decoding system based on OAK DSP Core. In order to solve the problem of resolution, the system has been used floating-point operation and double precision in dequantization module. The size of ROM is reduced by using the Run-length algorithm of reordered index. The subband synthesis filter module is optimized to have low computational complexity in terms of the size of ROM or RAM. To construct a efficient system, we used both the DSP Core and Parser-Huffman decoder which is implemented with VHDL.

• Journal of the Institute of Electronics Engineers of Korea TE
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• v.37 no.5
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• pp.56-61
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• 2000

Silicon semiconductor technology agrees that the number of transistors on a chip will keep growing exponentially, and it is pushing technology toward the System-On-Chip. In SoC Design, Specification at system level is key of success. Executable Specification reduces verification time. This Paper describes the design of IMDCT for MPEG Audio Decoder employing system-level design methodology and Executable Specification Methodology in the VHDL simulator with FLI environment.

• ETRI Journal
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• v.34 no.3
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• pp.474-477
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• 2012

The MPEG-D unified speech and audio coding (USAC) standardization process was initiated by MPEG to develop an audio codec that is able to provide consistent quality for mixed speech and music contents. The current USAC reference model structure consists of frequency domain (FD) and linear prediction domain (LPD) core modules and is controlled using a signal classifier tool. In this letter, we propose an LPD single-mode USAC structure using an adaptive widowing-based transform-coded excitation module. We tested our system using official test items for all mono-evaluation modes. The results of the experiment show that the objective and subjective performances of the proposed single-mode USAC system are better than those of the FD/LPD dual-mode USAC system.

• Journal of KIISE:Software and Applications
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• v.27 no.8
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• pp.864-876
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• 2000

This paper proposes a new scene boundary detection scheme for the MPEG System stream using MPEG Audio information and proves its usefulness by extensive experiments. A scene boundary has a characteristic that the audio as well as video information are changed rapidly. This paper first classifies this scene boundary into three cases ; Radical, Gradual, Micro Changes, with respect to the audio changes. The Radical change has a large-scale changing of decibel value and pitch value at a scene boundary, the Gradual change shows the long-time transition of decibel and pitch values from max to min or vice versa, and the Micro change displays a some change of pitch or frequency distribution without decibel changes. Upon this analysis, a new scene change detection algorithm detecting these three cases is proposed in which a progressive window with a time line is used to trace the changes in the audio information. Some experiments with various movies show that proposed algorithm could produce a high detection ratio for Radical change that is the most popular scene change in the movies, while producing a moderate detection ratio for Gradual and Micro changes. The proposed scene boundary detection scheme could be used to build a database for visual information like MPEG System stream.

• Journal of Broadcast Engineering
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• v.28 no.2
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• pp.213-229
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• 2023

In this paper, we introduce a spatial sound rendering system that provides 6DoF spatial sound in real time in response to the movement of a listener located in a virtual environment. This system was implemented using MPEG-I AEP as a development environment for the CfP response of MPEG-I Immersive Audio and consists of an encoder and a renderer including a decoder. The encoder serves to offline encode metadata such as the spatial audio parameters of the virtual space scene included in EIF and the directivity information of the sound source provided in the SOFA file and deliver them to the bitstream. The renderer receives the transmitted bitstream and performs 6DoF spatial sound rendering in real time according to the position of the listener. The main spatial sound processing technologies applied to the rendering system include sound source effect and obstacle effect, and other ones for the system processing include Doppler effect, sound field effect and etc. The results of self-subjective evaluation of the developed system are introduced.