• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1997.06a
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• pp.49-54
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• 1997

For the purpose of a flexible coded video transmission over a heterogeneous network, we propose a new packetization method for coded video data. The proposed method achieves small degradation of coded picture quality in case of packet discard at the network node and does not require heavy processing load for bitrate control operation. Computer simulation results show that the bitrate reduction from 384 kb/s to 192 kb/s does not cause severe degradation in picture quality.

• Journal of KIISE:Software and Applications
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• v.29 no.6
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• pp.355-366
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• 2002

This paper introduces the MPEG-4 video frame-based bitrate control methodology using two dimensional history pool and a sliding window. Proposed method preferentially clusters the encoded results according to the image characteristics and stores those results into the 2 dimensional history pool. Among the stored results in the pool, the sliding window collects the encoded results whose characteristics are very similar to the image frame to be encoded. Feedback regression is finally carried out bated on the collected results. Therefore proposed method can actively adapt to the rapid varying image characteristics by reducing the occurrences of the extrapolations when determining the quantization steps. Proposed method has better performances than the MPEG-4 frame-based bitrate control algorithm by evaluating with the actually encoded bits per frame, encoded PSNR's, and frame skips.

• 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.

• 한국HCI학회:학술대회논문집
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• 2008.02a
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• pp.218-223
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• 2008

In this paper, we implement several methods to improve an audio collaboration system over IP networks, and then evaluate the performance of the implemented methods. In general, speech and audio quality degrades depending on the characteristics of IP networks such as jitter and packet loss. In order to reduce this quality degradation, we propose a lower bit rate audio delivery scheme using the MPEG-2 AAC (Advanced Audio Coding) audio codec in a viewpoint that a packet loss rate could be reduced by a smaller packet size. In addition, iLBC (Internet Low-Bitrate Codec) and the G.711 packet loss concealment algorithm defined by IEFT and ITU-T, respectively, are applied to a audio collaboration system. RAT (Robust-Audio Tool)[7] is used as a baseline platform for the implementation of the proposed methods. It is shown from the implementation that the implemented MPEG-2 AAC audio codec with a bitrate of 256 kbit/s performs as similar as the uncompressed audio quality with a bitrate of 512 kbit/s, and that iLBC and the G.711 packet loss concealment algorithm can improve speech quality when a packet loss rate is 2~10%.

• 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.

• KIISE Transactions on Computing Practices
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• v.21 no.7
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• pp.458-463
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• 2015

These days, streaming users are using ABR (Adaptive Bitrate) technique services by requesting the most adequate video rate selectively based on their own channel states. Most ABR related video rate adaptation techniques are only focused on real-time bitrate adaptations based on their own channel state, and misses energy limited characteristics that come from a mobile device's battery dependence. In other words, the mobile device's important characteristics and accompanying energy consumption are not being considered and causes dissatisfaction over streaming services. In this paper, we propose energy efficient prefetching based dynamic adaptive video streaming techniques, which saves unnecessary consumed energy while providing video rates of the same performance. Our scheme continuously turns off energy modules with enough streaming in the buffer and turns on in case of the opposite situation to save energy. Through the performance evaluation, this study's proposed scheme is 60% better than the previous work at global average mobile download speed.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2016.10a
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• pp.969-972
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• 2016

In this paper, we propose an algorithm of angular mode selection for high-performance HEVC intra prediction. HEVC intra prediction is used to remove the spatial redundancy. Intra prediction has a total of 35 modes and block size of $64{\times}64$ to $4{\times}4$. Intra prediction has a high amount of calculation and operational time due to performing all 35 modes for each block size for the best cost. The angular mode algorithm proposed has a simple difference between pixels of the original image and the selected angular mode. A decision is made to select one angular mode plus planar mode and DC mode to perform the intra prediction and determine the mode with the best cost. In effect, only three modes are executed compared to the traditional 35 modes. Performance evaluation index used are BD-PSNR and BD-Bitrate. For the proposed algorithm, BD-PSNR results averagely increased by 0.035 and BD-Bitrate decreased by 0.623 relative to the HM-16.9 intra prediction. In addition, the encoding time is decreased by about 6.905%.

• 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 Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.4 no.1
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• pp.9-16
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• 2011

In this paper, we propose an efficience transcoder architecture to support a simutaneous multibitrate output. First, we discuss about some architectures to realize this feature. Next, we explain the proposed architecture, it shares not only VLD-IQ but alse Q-VlD which have the same quantization step sizes each other. We anlyze the numbers of Q-VLC times per on Macroblock to the investigate an effect of sharing, and evaluate its computation complexity. The result of simulation, that complexity has an upper limit and it cn support any numbers of bitstream by about 3~6 times complexity than single output.

• Journal of Korea Multimedia Society
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• v.3 no.3
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• pp.272-279
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• 2000

When the multiple video sources are together transmitted through the channel of fixed bandwidth, the efficiently joint bandwidth allocation is necessary. This paper presents the joint bandwidth allocation method to keep the constant picture duality ratio among the video sources. We first find an approximated model of distortion and bitrate for the multiplexing system of multiple sources. Then we obtain the bitrate for each source to have constant distortion ratio among the sources by using the approximated model parameters for simple implementation. It is shown by simulation that the proposed bandwidth allocation can keep almost constant picture quality ratio among the sources in comparison to an independent bandwidth allocation method.