• ETRI Journal
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• v.33 no.4
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• pp.641-644
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• 2011

In this letter, we propose a novel bitrate control, using both Kalman and FIR filters, based on a Hamiltonian analysis with respect to the amount of bits from each macroblock, in an encoding of a general video codec such as H.264/AVC. Since the proposed bitrate control is based on the simple computation of an optimal control method based on the Hamiltonian analysis, it is not necessary to use additional computation, such as a DCT or quantization, to estimate the bits for bitrate control. As a result, the proposed algorithm can be applied to single-pass encoding and can provide sufficient encoding speed with respect to various applications, even those requiring real-time control.

• Journal of Korea Multimedia Society
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• v.9 no.7
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• pp.818-826
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• 2006

In this paper, we investigate the relationship of bitrate and quantization parameter needed for the trans-coding method that makes the H.264 bitstream of a particular bitrate to the other titrate. Also we propose the new method in order to transcode the titrate between H.264 video coded bitstreams. The proposed transcoding method updates the model parameters from previous picture or slice by using the approximated relationship of bitrate and quantization step-size and finds the target quantization step-size, and then generates the target titrate by simple coding processing just after requantization. Therefore, the proposed method does not need the complex bitrate control and converts to the target titrate by simple implementation. From simulation, we can see that the proposed method transcodes exactly to an assigned target bitrate for the four test sequences with different their characteristics.

• The Journal of the Acoustical Society of Korea
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• v.24 no.6
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• pp.358-364
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• 2005

In this Paper. we present a bitrate scalable speech codec which uses an ITU-T G.723.1 as the baseline coder and encodes the synthesis error signal in an enhancement coder. ITU-T P.862 (PESQ) is used to evaluate the Performance of the bitrate scalable coder. Experiments show that 6.7kbps enhancement layer based on G.723.1 5.3kbps produces the increase of 0.39 in MOS and 5.7kbps enhancement layer based on G.723.1 6.3kbps Produces the increase of 0.267 in MOS.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.4
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• pp.1877-1891
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• 2016

In video coding, the goal of rate control (RC) is not only to avoid the undesirable fluctuation in bit allocation, but also to provide a good visual perception. In this paper, a novel frame-level rate control algorithm for High Efficiency Video Coding (HEVC) is proposed. Firstly a model that reveals the relationship between bit per pixel (bpp), the bitrate of the intra frame and the bitrate of the subsequent inter frames in a group of pictures (GOP) is established, based on which the target bitrate of the first intra frame is well estimated. Then a novel frame-level bit allocation algorithm is developed, which provides a robust bit balancing scheme between the intra frame and the inter frames in a GOP to achieve the visual quality smoothness throughout the whole sequence. Our experimental results show that when compared to the RC scheme for HEVC encoder HM-16.0, the proposed algorithm can produce reconstructed frames with more consistent objective video quality. In addition, the objective visual quality of the reconstructed frames can be improved with less bitrate.

• ETRI Journal
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• v.27 no.2
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• pp.231-234
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• 2005

This letter proposes a new embedded speech coding structure based on the Adaptive Multi-Rate Wideband (AMR-WB) standard codec. The proposed coding scheme consists of three different bitrates where the two lower bitrates are embedded into the highest one. The embedded bitstream was achieved by modifying the algebraic codebook search procedure adopted for the AMR-WB codec. The proposed method provides the advantage of scalability due to the embedded bitstream, while it inevitably requires some additional computational complexity for obtaining two different code vectors of the higher bitrate modes. Compared to the AMR-WB codec, the embedded coder shows improved speech qualities for two higher bitrate modes with a slightly increased bitrate caused by the decreased coding efficiency of the algebraic codebook.

• Journal of information and communication convergence engineering
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• v.8 no.3
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• pp.245-250
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• 2010

To support various bandwidth requirements for many kinds of devices such as PC, notebook, PDA, cellular phone, a transcoding proxy is usually necessary to provide not only adapting multimedia streams to the client by transcoding, but also caching them for later use. Due to huge size of streaming media, we proposed the 3 kinds of segmentation - PT-2, uniform, bitrate-based segmentation. And to reduce the CPU cost of transcoding video, we proposed the DQS service policy. In this paper, we simulate the combined our previous two researches that are bitrate-based segmentation and DQS(Degraded Quality Service) policy. Experimental results show that the combined policy outperforms companion schemes in terms of the byte-hit ratios and delay saving ratios.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.7
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• pp.1-5
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• 1998

In this paper, recovery method with high-bitrate and low-bitrate coder was implemented in order to recover consecutive packet loss over the Internet. LPC was used as redundant audio data for recover of lost packets and RTP parcket format was modified for accommodation of redundant data. In measuring results using random packet loss rate with three redundant datra in every packet, it has shown that recovery rate was 80% in los rate of 50%. Since the processing delay for recovery of the lost packet was 200ms, this recovery method can be applied to real-time Internet sevice such as Internet phone.

• Journal of Communications and Networks
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• v.15 no.6
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• pp.635-644
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• 2013

Adaptive hypertext transfer protocol (HTTP) streaming has become a new trend to support adaptivity in video delivery. An HTTP streaming client needs to estimate exactly resource availability and resource demand. In this paper, we focus on the most important resource which is bandwidth. A new and general formulation for throughput estimation is presented taking into account previous values of instant throughput and round trip time. Besides, we introduce for the first time the use of bitrate estimation in HTTP streaming. The experiments show that our approach can effectively cope with drastic changes in connection throughput and video bitrate.

• Journal of Broadcast Engineering
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• v.17 no.2
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• pp.325-343
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• 2012

This paper proposes the coding method to predict the split structure of LCU in the current frame on the basis of the reference frame or temporally-previous frame. HEVC encoder determines split structure according to image characteristics in LCU which is an basic element of CU. The split structure of the current LCU is very similar to the split structure of collocated LCU in the reference frame or temporally-previous frame. Thus, this paper proposes the method to reduce the encoder computational complexity by predicting split structure of the current LCU on the basis of that of collocated LCU in the reference frame or temporally-previous frame. And it also proposes the method to reduce the BD-Bitrate by coding after the prediction of the CU split information. The simulation results of changing only encoder showed that the mean of encoder computational complexity was lower by 21.3%, the decoder computational complexity was negligible change and the BD-Bitrate increase by the maximum of 0.6%. Also, the method changing encoder, bitstream, and decoder improves the mean of encoder computational complexity was lower by 22%, the decoder computational complexity was negligible change and the BD-Bitrate is improved to the maximum of 0.3%. When compared with the conventional method, indicating that the proposed method is superior.

• Journal of Broadcast Engineering
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• v.23 no.6
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• pp.768-779
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• 2018

In this paper, we propose a novel video quality function for video processing based on MSSSIM to select an appropriate video processing filter and to accommodate multiple processing filters to each pixel block in a picture frame by a mathematical selection law so as to maintain video quality and to reduce the bitrate of compressed video. In viewpoint of video compression, since the properties of video quality and bitrate is different for each picture of video frames and for each areas in the same frame, it is difficult for the video filter with single property to satisfy the object of increasing video quality and decreasing bitrate. Consequently, to maintain the subjective video quality in spite of decreasing bitrate, we propose the methodology about the MSSSIM as the measure of subjective video quality, the KL-Divergence as the measure of bitrate, and the combination method of those two measurements. Moreover, using the proposed combinatorial measurement, when we use the multiple image filters with mutually different properties as a pre-processing filter for video, we can verify that it is possible to compress video with maintaining the video quality under decreasing the bitrate, as possible.