• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.5
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• pp.147-154
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• 2015

In this paper, we employed a model which defines the sensitivity according to the background luminance, so called JND (Just Noticeable Difference), and applied to the video coding. The proposed method finds out the maximum possible quantization parameter for the current unit based on the threshold of JND model and reduce the bitrate with similar perceptual quality. It selects the higher quantization parameter and reduce the bitrate when the reconstructed signal which is coded with higher quantization parameter is in a range of allowance based on the JND threshold, i.e. the signal has the similar perceptual quality compared to that is coded with the initial quantization parameter. The proposed algorithm was implemented on HM16.0, which is a reference software of the latest video coding standard HEVC (High Efficiency Video Coding) and the coding performance was evaluated. Compared to HM16.0, the proposed algorithm achieved maximum 20.21% and 6.18% of average bitrate reduction with the similar perceptual quality.

• Journal of Broadcast Engineering
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• v.23 no.1
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• pp.104-114
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• 2018

Recently, as HTTP-based video streaming traffic continues to increase, Dynamic Adaptive Streaming over HTTP(DASH), which is one of the HTTP-based adaptive streaming(HAS) technologies, is receiving attention. Accordingly, many video quality control techniques have been proposed to provide a high quality of experience(QoE) to clients in a DASH environment. In this paper, we propose a new quality control method using ANFIS(Adaptive Network based Fuzzy Inference System) which is one of the neuro-fuzzy system structure. By using ANFIS, the proposed scheme can find fuzzy parameters that selects the appropriate segment bitrate for clients. Also, considering the characteristic of VBR video, the next segment download time can be more accurately predicted using the actual size of the segment. And, by using this, it adjusts video quality appropriately in the time-varying network. In the simulation using NS-3, we show that the proposed scheme shows higher average segment bitrate and lower number of bitrate-switching than the existing methods and provides improved QoE to the clients.

• The Journal of the Acoustical Society of Korea
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• v.22 no.2
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• pp.129-134
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• 2003

In this paper, we efficiently implement the AMR speech coder by reducing the complexity of algebraic codebook search. To reduce the computational complexity of the algebraic codebook search, we propose a fast algebraic codebook search method that improves conventional depth first tree search method used in AMR speech coder algorithm. The proposed method reduces the search complexity by pruning the trees which are less possible to be selected as an optimum excitation. This method needs no additional computation for selecting the trees to be pruned and reduces the computational complexity considerably compared to the original depth first tree search method with slightly degradation or speech qualify. Applying our method to the implementation or AMR speech coder with 12.2 kbps mode by using the TeakLite DSP, we reduce the search complexity about 40% compared to the conventional method.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.8
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• pp.86-95
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• 1998

Intra-frame video compression, which cannot make use of temporal predictions, requires much higher bit rates compared with inter-frame schemes. In order to reduce bit rates, intra-frame predictive coding methods in DCT domain have been studied especially within the framework of the MPEG-4 video coding standard currently being developed. In this paper, we propose novel intra-frame predictive coding methods in DCT domain with the marginal complexity increase over the conventional methods . The proposed methods consist of a DC coefficient prediction method and two AC coefficient prediction methods. The proposed methods consist of a DC coefficient prediction method and two AC coefficient prediction methods. The proposed DC coefficient prediction method makes it possible to adaptively select the prediction directions without overhead bits, by comparing gradients of DC coefficients from neighboring blocks. As the AC coefficient prediction methods, first, we present an effective method which can improve the prediction directions of the MPEG-4 scheme by considering the DC coefficient of the current block to be coded. And, we present another effective method that decision on the prediction is carried out for each AC coefficient. Simulation results show that substantial bit savings can be achieved by the proposed methods.

• The Journal of the Acoustical Society of Korea
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• v.19 no.4
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• pp.37-44
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• 2000

In this paper, we described the real-time implementation effort of MPEG-4 audio HVXC (Harmonic Vector eXcitation Coding) algorithm for very low bitrates, which has target applications from mobile communications to Internet telephony, on current high performance floating point TMS320C6701 DSP. We adopted a hardware structure for real-time operation. In order for software optimization, we used C- and assembly-language level optimizations for time-critical functional codes. Utilizing the internal program memory of the DSP as the program cache, the internal data memory overlap technique and DMA functionality, we could get a goal of realtime operation of HVXC codec both at 2 kbit/s and at 4 kbit/s. For an encoder at 2 kbit/s, the optimization ratio to original code is about 96 %. Finally, we got the subjective quality of MOS 2.45 at 2 kbit/s from an informal quality test.

• 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.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2012.07a
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• pp.428-430
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• 2012

Korea View 는 지상파 디지털 TV 의 새로운 서비스 프레임워크로 6MHz 의 지상파 방송 대역폭 내에서 하나의 HD 채널만 서비스되던 것을 향상된 비디오 압축 기술을 이용하여 1 개의 HD 채널과 3 개의 SD 채널, 총 4 개의 채널을 서비스하는 지상파 다채널 디지털 TV 서비스를 의미한다. 본 논문에서는 Korea View 서비스 프레임워크에 H.264/SVC 를 이용하여 SD 급으로 전송되던 3 개의 채널에 대해 SVC 의 공간 계위를 갖도록 부호화 한 HD 급 향상계층 비디오 부호화 스트림을 광대역 망을 통해 전송하는 다채널 하이브리드 HDTV 방송서비스 프레임워크를 제안하고 이를 위해 H.264/SVC 의 부호화 성능을 평가하고 분석한다. 이를 위해 본 논문에서는 H.264/SVC 기본계층에 해당하는 3 개의 SD 채널에 대해 1.5Mbps 목표 비트율로 부호화 하는 동시에 H.264/SVC 의 향상계층에 해당하는 HD 비디오의 부호화를 수행하여 SD 급 기본계층과 HD 급 향상계층의 부호화 성능을 평가 및 분석하였다. 실험을 통해 SD 급 기본계층에서 34.5dB 이상의 화질로 서비스 가능한 수준의 객관적 화질임을 확인하였고, HD 급 향상계층에 대해서는 12Mbps 비트율로 부호화 된 MPEG-2 영상의 화질을 제공하기 위해 690kbps 이상의 비트율이 필요하다는 것을 제시한다.

• The Journal of the Korea Contents Association
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• v.15 no.2
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• pp.1-10
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• 2015

In this paper, we propose an integral image compression method to improve the coding efficiency. As the characteristics of integral images are various according to the distance between a camera and objects, complexity of the objects and background, etc, the coding efficiency can be improved by applying a coding method adaptive to the characteristics. In addition, as the integral images are compressed by the unit of elemental images, the coding efficiency can be improved. by employing a coding method optimized to the coding direction of elemental images. Therefore, the proposed method remaps an integral image with six kinds of mapping rules, and then the conventional 3D-DCT based compression method is applied to the remapped images. Finally, we perform the rate-distortion optimization to choose the best of the mapping rules. Experimental results show that the proposed method yields high gains in image quality and bit-rate.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.42 no.5 s.305
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• pp.115-122
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• 2005

There are some problems in directly applying the conventional MPEG bit-stream based watermarking schemes to the bit-stream of a new compression standard, H.264/AVC. In this paper we analyze the effects of the conventional DCT-based watermarking scheme to H.264/AVC, especially in terms of image distortions and bit-rate changes. It turns out that the intra-frame prediction md CAVLC of H.264/AVC with the watermarking worsen the image distortions and bit-rate changes. The experiment results show on average 28.17dB decrease in PSNR and 56.71% increase in bit-rate over all QPs.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.4C
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• pp.312-320
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• 2008

The macroblock-based adaptive interpolation filter method has been considered to be able to achieve high coding efficiency in H.264/AVC. In this method, although the filter selection criterion considered in terms of rate and distortion have showed a good performance, it still leaves room for improvement. To improve high coding efficiency better than conventional method, we propose a new filter selection criterion which considers two bit rates, motion vector and prediction error, and reconstruction error. In addition, the algorithm for reducing the overhead of transmitting the selected filter information is presented. Experimental results show that the proposed method significantly improves the coding efficiency compared to ones using conventional criterion. It leads to about a 5.19% (1 reference frame) and 5.14% (5 reference frames) bit rate savings on average compared to H.264/AVC, respectively.