• Journal of KIISE
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• v.44 no.1
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• pp.95-106
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• 2017

HTTP adaptive streaming has recently emerged to handle the rapidly growing traffic and to provide high quality multimedia contents. To improve the QoE (Quality of Experience) for HTTP adaptive streaming service, the average video bitrate should be maximized, and the video switching frequency (difference of bitrate between adjacent segments) and video stalling events need to be minimized. The recently proposed quality adaptation algorithms for HTTP adaptive streaming do not provide high QoE, since detailed QoE factors such as video switching frequency and bitrate difference of adjacent segments, are not considered. In this paper, we propose a SQA (Smooth Quality Adaptation) algorithm to improve the user QoE. The proposed algorithm provides the smoothed QoE, such that it minimizes the unnecessary video switching events by maintaining the quality in a certain period, thus minimizing the bitrate difference of adjacent segments. Through simulation, we confirm that the proposed algorithm reduces the unnecessary switching events, and prevents the sudden decrease in video quality.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.305-308
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• 2000

IP 네트웍을 통해 실시간 비디오를 전송할 때 네트윅 특성을 고려하여 비디오 데이터를 부호하면 패킷 손실로 인한 품질 열화를 최소화하여 더 좋은 품질의 비디오를 얻을 수 있다. 이를 위해 현재 네트윅의 정보를 빠르고 정확하게 얻어내는 메커니즘과 부호화 변수를 네트윅 상황에 적응적으로 조절하여 패킷 손실에 강인한 압축 및 전송 메커니즘의 두 가지 기능이 요구된다. 첫번째 메커니즘은 RTP(Real Time Transport Protocol)을 통해 구현될 수 있으며, 두 번째 메커니즘을 위해 본 연구에서는 다중기술 변환부호화(Multiple Description Transform Coding) 기법을 적용한 비디오 부호화 알고리즘을 제안한다. RTP에서 제공하는 RTCP(Real Time Control Protocol) 정보를 이용하여 현재 네트웍 정보를 얻을 수 있으며, 다중기술 변환부호화 기법을 이용하여 현재의 패킷 손실률에서 최적의 품질을 보장하도록 부호화 변수를 조절할 수 있다. 본 논문에서는 다중기술 변환부호화 기법을 비디오 부호화에 적용하여 순수 비디오 정보에 추가되는 잉여 정보량과 패킷 손실에 대한 강인성 사이의 관계를 도출함으로써 다중기술 변환부호화 기법이 네트웍 적응적 부호화 방식에 적합한 방식임을 제시한다.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.10 no.12
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• pp.2315-2320
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• 2006

A real-time frame-layer rate control algorithm with a token bucket traffic shaper is proposed for low bit rate video coding. The proposed rate control method uses a non-iterative optimization method for low computational complexity, and performs bit allocation at the frame level to minimize the average distortion over an entire sequence as well as variations in distortion between frames. In order to reduce the quality fluctuation, we use a sliding window scheme which does not require the pre-analysis process. Therefore, the proposed algorithm does not produce time delay from encoding, and is suitable for real-time low-complexity video encoder. Experimental results indicate that the proposed control method provides better visual and PSNR performances than the existing rate control method.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2014.06a
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• pp.203-205
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• 2014

본 논문에서는 HEVC 기반 비디오 부호화에 있어 CTU 단위의 시각 민감도에 따라 CTU 별로 QP 를 조절하여 주관적 화질을 향상시키는 방법을 제안한다. 시각 민감도를 측정하는 방법으로서 화소 영역에서의 최소가지차(JND, just noticeable distortion)를 계산하여 이용하였고, 이를 HM 12.0 참조 소프트웨어에서 이용되는 $R-{\lambda}$ 모델 기반의 율 제어 모듈에 결합하여 시각 민감도에 따라 QP 를 제어할 수 있도록 하였다. 시각 민감도가 큰 영상의 영역에 대해서는 상대적으로 작은 QP 값을, 시각민감도가 작은 영역에 대해서는 큰 QP 값을 양자화 과정에 적용함으로써, 시각 민감도가 작은 영역에 대해서는 사용 비트양을 절약하고, 절약된 비트를 상대적으로 시각 민감도가 큰 영역을 위해 사용함으로써 비디오의 주관적 화질을 향상시킬 수 있었다. 뿐만 아니라 이를 하드웨어에 적용 가능하게 하기 위해 HM 12.0 기반 하드웨어 구현을 위한 소프트웨어 플랫폼에 구현하여 실험한 결과, $R-{\lambda}$ 모델 율 제어 알고리즘으로 율 제어 하여 부호화 한 경우 Y-PSPNR(peak signal to perceptual noise ratio)에 대한 BD-rate 는 평균 9.4%의 이득이 있었음을 확인하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4B
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• pp.202-209
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• 2005

In this paper, a dynamic TCP-friendly rate control (TFRC) is proposed to adjust the coding rates according to the channel characteristics of the wireless-to-wired network consisting of wireless first-hop channel. To avoid the throughput degradation of multimedia flows traveling through wireless lint the proposed rate control system employs a new wireless loss differentiation algorithm (LDA) using packet loss statistics. This method can produce the TCP-friendly rates while sharing the backbone bandwidth with TCP flows over the wireless-to-wired network. Experimental results show that the proposed rate control system can eliminate the effect of wireless losses in flow control of TFRC and substantially reduce the abrupt quality degradation of the video streaming caused by the unreliable wireless link status.

• Journal of KIISE:Computer Systems and Theory
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• v.33 no.10
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• pp.722-733
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• 2006

In scalable streaming application, there are two important knobs to tune to effectively exploit the underlying network resource and to maximize the user perceivable quality of service(QoS): layer selection and packet scheduling. In this work, we propose Semantics Aware Packet Scheduling (SAPS) algorithm to address these issues. Using packet dependency graph, SAPS algorithm selects a layer to maximize QoS. We aim at minimizing distortion in selecting layers. In inter-frame coded video streaming, minimizing packet loss does not imply maximizing QoS. In determining the packet transmission schedule, we exploit the fact that significance of each packet loss is different dependent upon its frame type and the position within group of picture(GOP). In SAPS algorithm, each packet is assigned a weight called QoS Impact Factor Transmission schedule is derived based upon weighted smoothing. In simulation experiment, we observed that QOS actually improves when packet loss becomes worse. The simulation results show that the SAPS not only maximizes user perceivable QoS but also minimizes resource requirements.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• Journal of the Institute of Electronics and Information Engineers
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• v.51 no.12
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• pp.20-29
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• 2014

Recently, LTE is preparing to make a new leap forward LTE-A all over the world. As LTE privides high speed service, the role of mobile phones seems to change from voice to data service. According to Cisco, global mobile data traffic will increase nearly 11-fold between 2013 and 2018. Mobile video traffic will reach 75% by 2018 from 66% in 2013 in Korea. However, voice service is still the most important role of mobile phones. Thus, controllability of throughput and low BLER is indispensable for high-quality VoIP service among various type of traffic. Although the maximum AMR-WB, 23.85 Kbps is sufficient to a VoIP call, it is difficult for the LTE which can provide tens to hundreds of MB/s may not keep the certain level VoIP QoS especially in the cell-edge area. This paper proposes a new scheduling algorithm in order to improve VoIP performance after analyzing various scheduling algorithms. The proposal is the technology which applies more priority processing for VoIP than other applications in cell-edge area based on two-tier scheduling algorithm. The simulation result shows the improvement of VoIP performance in the view point of throughput and BLER.