• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.822-829
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• 2009

This paper proposes a new method of improving the quality of High Efficiency Advanced Audio Coding (HE-AAC). HE-AAC encodes input source by allocating bits for each scalefactor bands appropriately according to human ear's psychoacoustic property. As a result, insufficient bits are assigned to the bands which have relatively low energy. This imbalance between different energy bands can cause decreasing of sound quality like musical noise. In the proposed system, a Linear Prediction (LP) module is combined with HE-AAC as a pre-processor to improve sound quality by even bits distribution. To apply accurate human being's psychoacoustic property, the psychoacoustic model uses Fast Fourier Transform (FFT) spectrum of original input signal to make masking threshold. In its implementation, masking threshold of psychoacoustic model is normalized using the LP spectral envelope in prior to quantization of the LP residual. Experimental result shows that, the proposed algorithm allocates bits appropriately for insufficient bits condition and improves the performance of HE-AAC.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.6
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• pp.341-349
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• 2014

Dynamic adaptive streaming over HTTP (DASH) is the most recent and promising technology to support high quality streaming services. In dynamic adaptive streaming over HTTP (DASH), a client consecutively estimates the available network bandwidth and decides the transmission rate for the forthcoming video chunks to be downloaded. In this paper, we propose a novel rate adaptation algorithm called quality of experience QoE-enhanced adaptation algorithm over DASH (QAAD), which preserves the minimum buffer length to avoid interruption and minimizes the video quality changes during the playback. We implemented a DASH test bed and conducted extensive experiments. Experimental results demonstrate that under fluctuating network conditions, QAAD provides seamless streaming with stabilized video quality while the previous buffer-aware algorithm (i.e., QDASH[9]) frequently changes the video quality and undergoes the interruption.

• Journal of Broadcast Engineering
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• v.22 no.3
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• pp.381-396
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• 2017

In this paper, the perceptual rate control algorithm is studied for HEVC (High Efficiency Video Coding) encoder with bit allocation based on perceived visual quality. This paper proposes perceptual rate control algorithm which could consider perceived quality for HEVC encoding method. The proposed rate control algorithm employs adaptive bit allocation for frame and CTU level using the perceived visual importance of each CTU. For performance evaluation of the proposed algorithm, the proposed algorithm was implemented on HM 16.9 and tested for sequences in Class B under the CTC (Common Test Condition) RA (Random Access) case. Experimental results show that the proposed method reduces the bitrate of 3.12%, and improves BD-PSNR of 0.08dB and bitrate accuracy of 0.07% on average. And also, we achieved MOS improvement of 0.16 with the proposed method, compared with the conventional method based on DSCQS (Double Stimulus Continuous Quality Scale).

• 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

• Journal of Internet Computing and Services
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• v.17 no.6
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• pp.9-15
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• 2016

In this paper, we propose a smart HLS(HTTP Live Stream) platform with dynamic single buffering for the best transmission of adaptive video bit-rates. This smart HLS can optimizes the channel transition zapping-time with the monitoring of bandwidth between HLS server and OTT(Over The Top) client. This platform is designed through the control of video stream due to proper multi-bitrates and bandwidths. This proposed OTT can decode the live and VOD(Video On Demand) videos with the buffering of optimumal bitrate. And, the HLS can be cooperated with a smart OTT, and segmented for the m3u8 files of H.265 MPEG-2 TS(Transport Stream) videos. As a resullt, this single buffer based smart OTT can transmit optimal videos with the maximum data buffering according to the adaptive bit-rate depending on the network bandwidth efficiency and the decoded VOD video, too.

• The Journal of the Acoustical Society of Korea
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• v.27 no.2
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• pp.66-74
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• 2008

In this paper, we propose new techniques that can improve the quality of AAC and SBR encoders comprised in low bitrate HE-AAC. To reduce the pre-echo artifacts often occurring for transient blocks in AAC, we propose an extended Temporal Noise Shaping (sTNS) in which the frequency range is selectively extended down to the low-frequency region. Also, for he high-frequency region being coded by SBR encoder, tones are identified through a sinusoidal modeling and their frequencies are adjusted within the QMF band in order to reduce the noise floor due to aliasing. Spectrograms of the decoded signals were compared and listening tests were conducted to evaluate the proposed algorithm. Results confirmed the effectiveness of the proposed algorithm.

• Information and Communications Magazine
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• v.11 no.8
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• pp.60-77
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• 1994

This paper presents a survey of video coding schemes which use wavelet transform for the videophone on very low bit rate commun ication chan nel( ego 10 Kbps Public Service Telephone Network). Firstly, we introduce the standardization efforts to make the low bit rate videophone architecture and the typical application of low bit rate video coding scheme. Secondly, we summarize the several requirements on videophone, delay, encoder/decoder complexity, low bitrate, and progressive transmission capability. Third, we review the basic theory of wavelet transform without much mathematics. We compare the wavelet transform with short-time fourier transform and subband filters. Fourth, we summarize the video coding schemes proposed so far, and evaluate them with Ule requirements. Lastly, we conclude with fu¬ture research directions.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.3
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• pp.1093-1104
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• 2014

The High Efficiency Video Coding (HEVC) is a new video coding standard that can provide much better compression efficiency than its predecessor H.264/AVC. However, it is computationally more intensive due to the use of flexible quadtree coding unit structure and more choices of prediction modes. In this paper, a fast intraframe coding scheme is proposed for HEVC. Firstly, a fast bottom-up pruning algorithm is designed to skip the mode decision process or reduce the candidate modes at larger block size coding unit. Then, a low complexity rough mode decision process is adopted to choose a small candidate set, followed by early DC and Planar mode decision and mode filtering to further reduce the number of candidate modes. The proposed method is evaluated by the HEVC reference software HM8.2. Averaging over 5 classes of HEVC test sequences, 41.39% encoding time saving is achieved with only 0.77% bitrate increase.

• ETRI Journal
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• v.31 no.5
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• pp.610-612
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• 2009

We present intra-mixture prediction (IMP) mode for intra prediction and an enhanced estimation method for most probable mode (MPM). IMP mode supports more flexibility in intra prediction by mixing $4{\times}4$ blocks and $8{\times}8$ blocks in one macroblock, while the enhanced MPM estimation extends the number of referenced neighboring blocks and efficiently uses their prediction modes depending on their positions. Simulation results show that the combination of both proposed methods provides a bit reduction in the Bj${\phi}$ntegaard delta bitrate by an average of 2.56% compared to H.264/AVC.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.1105-1108
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• 2005

Media service model is divided into 3 models that are Client Pull, Server Push, and IPP(Interleaving Pull & Push) model. In most single VOD(Video On Demand) environment, Client Pull model was sufficient to play the movie Because most media contents has a low bitrate and resolution. But according to an increment of the demand of the high definition media, Client Pull model is not sufficient. Parallel VOD environment is made of several of VOD servers and provides the parallel media stream simultaneously for one client. We compared and analyzed the performance of service models with respect to network delay and data size in buffer in the single and parallel VOD environment and we found that IPP service model keeps the least network delay and stable client buffer state in the parallel VOD environment.