• Journal of Broadcast Engineering
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• v.10 no.4 s.29
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• pp.488-504
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• 2005

In this paper, we propose a new bitrate control scheme to improve the quality of image encoded by SVC and to resolve the problems of conventional scheme. In JSVM2.0, bitrate of a frame is controlled by an initial quantization parameter and scaling factor that it hasdifferent value according to frame. Itis difficult to get the best of video quality at arbitrary bitrate because the conventional scheme has two defects. One is that we have to know proper initial QP's fur all sequences. Another is that QP's control skill for macroblocks is very inefficient. In this paper, we propose an efficient bit allocation algorithm to reduce the effect of the initial QP and to increase the efficiency of bit allocation by using proper QP's for macroblocks. In simulation results, it can be seen that using the proposed scheme enables the SVC encoder to control the bitrate by the macroblock unit and outperforms the conventional schemes in the respect of rate-distortion.

• 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 Korea Multimedia Society
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• v.8 no.6
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• pp.744-752
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• 2005

When the multiple video sources are transmitted together through the channel of fixed bandwidth, an efficient picture quality control is necessary. This paper presents a joint picture quality control method to satisfy the user requirement by priority among the video sources. The better picture quality is assigned to the source of higher priority compared to lower priority. We obtain the bitrate for each source to have a required distortion level among the sources by using an approximate distortion-bitrate model for simple implementation. It is shown by simulation that the proposed bandwidth allocation method can keep almost the user-required picture quality among the sources in comparison to an independent bandwidth allocation method.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.5
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• pp.303-309
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• 2016

This paper presents a channel-adaptive rate control algorithm for low delay video coding. The main goal of the proposed method is to adaptively use the unknown available channel bandwidth while reducing the end-to-end delay between encoder and decoder. The key idea of the proposed algorithm is for the status of the encoder buffer to indirectly reflect the mismatch between the available channel bandwidth and the generated bitrate. Hence, the proposed method fully utilizes the unknown available channel bandwidth by monitoring the encoder buffer status. Simulation results show that although the target bitrate mismatches the available channel bandwidth, the encoder efficiently adapts the given available bandwidth to improve the peak signal-to-noise ratio.

• Journal of the Institute of Electronics and Information Engineers
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• v.50 no.8
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• pp.245-249
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• 2013

In this paper, we propose two rate control algorithms for multi-view extension of HEVC with two rate control algorithms adopted in HEVC and analyze the multi-view rate control performance. The proposed multi-view rate controls are designed on HEVC-based multi-view video coding (MV-HEVC) platform with consideration of high-level syntax, inter-view prediction, etc. not only for the base view but also for the extended views using the rate control algorithms based on URQ (Unified Rate-Quantization) and R-lambda model adopted in HEVC. The proposed multi-view rate controls also contain view-wise target bit allocation for providing the compatibility to the base view. By allocating the target bitrates for each view, the proposed multi-view rate control based on URQ model achieved about 1.83% of average bitrate accuracy and 1.73dB of average PSNR degradation. In addition, about 2.97% of average bitrate accuracy and 0.31dB of average PSNR degradation are achieved with the proposed multi-view rate control based on R-lambda model.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.2A
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• pp.130-136
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• 2002

It is required to control the picture quality among video sources when the multiple video sources are together transmitted through the channel of fried bandwidth such as satellite and terrestrial broadcasting. This paper presents a control method to keep the uniform picture quality among the video sources. We first find a model of distortion and titrate far the multiplexing system of multiple sources. Then we allocate the bitrate to each source for uniformly divided picture quality among the sources by using the distortion-bitrate relationship for the real-time implementation without time delay. It is shown by simulation that the proposed quality control method can keep almost uniform picture quality among the sources in comparison to the conventional control method.

• ETRI Journal
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• v.38 no.6
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• pp.1064-1073
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• 2016

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

• Journal of Broadcast Engineering
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• v.10 no.3
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• pp.348-356
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• 2005

In this paper, we propose a fast bitrate reduction transcoding method to convert a bitstream coded by H.264 into a lower bitrate H.264 bitstream. Block mode informations and motion vectors generated by H.264 decoder are used for probability-based block mode determination in the proposed transcoding method. And the motion vector reuse and motion vector refinement process are applied in the proposed transcoding. In the experiment results, the proposed methods achieves approximately 40 times improvement in computation complexity compared with the cascaded pixel domain transcoding, while the PSNR(Peak Signal to Noise Ratio) is degraded with only $0.1\~0.3$ dB.

• 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.

• The Transactions of the Korean Institute of Electrical Engineers D
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• v.51 no.7
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• pp.302-307
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• 2002

Digital watermarking is the technique, which embeds an invisible signal including signal including owner identification and copy control information into multimedia data such as audio, video, and images for copyright protection. A new MPEG watermark embedding algorithm using complex block effect based on the Human Visual System(HVS) is introduced in this paper. In this algorithm, $8{\times}8$ dark blocks are selected, and the watermark is embedded in the DC component of the discrete cosine transform(DCT) by using quantization and modulus calculation. This algorithm uses a blind watermark retrieval technique, which detects the embedded watermark without using the original image. The experimental results show that the proposed watermark technique is robust against MPEG coding, bitrate changes, and various GOP(Group of Picture) changes.