• Proceedings of the Korea Information Processing Society Conference
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• 2004.05a
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• pp.1319-1322
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• 2004

스무딩은 가변 비트율로 저장된 비디오 데이터를 클라이언트로 전송할 때 일련의 고정 비트율로 전송할 수 있도록 전송 계획을 세우는 것이다. 이러한 스무딩 알고리즘에는 CBA, MCBA, MVBA, PCRTT, e-PCRTT등이 있다. 특히, PCRTT 알고리즘의 문제점을 개선한 e-PCRTT 알고리즘은 전송률 변화 횟수가 주어지고 구간의 크기가 고정적인 특징을 갖고 있어 전송률 변화 횟수, 첨두 전송률, 버퍼 이용률등의 평가 요소들이 증가될 수 있다. 따라서, 본 논문에서는 e-PCRTT 알고리즘의 문제점을 해결하기 위해 전송률 변화 횟수의 제한이 없고 구간의 크기가 가변적인 스무딩 알고리즘을 제안한다. 제안 알고리즘의 성능 평가를 위해 3개의 비디오 소스를 사용하여 전송률 변화 횟수, 첨두 전송률, 그리고 버퍼 이용률을 비교한다. 제안 알고리즘은 전송률 변화 횟수, 첨두 전송률은 e-PCRTT 알고리즘 보다 우수하고, 버퍼 이용률 비교에서는 비슷한 결과를 보인다.

• Journal of IKEEE
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• v.2 no.2 s.3
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• pp.233-238
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• 1998

In this paper, the implementation of dual rate ADPCM using G.726 16Kbps and 40Kbps speech codec algorithm is handled. For small signals, the low rate 16Kbps coding algorithm shows almost the same SNR as the high rate 40Kbps coding algorithm , while the high rate 40Kbps coding algorithm shows the higher SNR than the low rate 16Kbps coding algorithm fur large signal. To obtain the good trade-off between the data rate and synthesized speech quality, we applied low rate 16Kbps for the small signal and high rate 40Kbps for the large signal. Various threshold values determining the rate are applied for good trade-off between data rate and speech quality. The simulation result shows the good speech quality at a low rate comparing with 16Kbps & 40Kbps.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.9C
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• pp.849-860
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• 2007

Because the H.264/AVC standard adopts the variable length coding algorithm, the rate of encoded video bitstream fluctuates a lot as time flows, though its compression efficiency is superior to that of existing standards. When a video is transmitted in real-time over networks with fixed low-bandwidth, it is necessary to control the bit rate which is generated from encoder. Many existing rate control algorithms have been adopting the quadratic rate-distortion model which determines the target bits for each frame. We propose a new rate control algorithm for H.264/AVC video transmission over networks with fixed bandwidth. The proposed algorithm predicts quantization parameter adaptively to reduce video distortion using the quadratic rate-distortion model, which uses the target bit rate and the mean absolute difference for current frame considering pixel difference between macroblocks in the previous and the current frame. On video samples with high motion and scene change cases, experimental results show that (1) the proposed algorithm adapts the encoded bitstream to limited channel capacity, while existing algorithms abruptly excess the limit bit rate; (2) the proposed algorithm improves picture quality with $0.4{\sim}0.9dB$ in average.

• Information and Communications Magazine
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• v.23 no.7
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• pp.108-122
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• 2006

• The KIPS Transactions:PartC
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• v.9C no.5
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• pp.679-686
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• 2002

The transmission of prerecorded and compressed video data without degradation of picture quality requires video servers to cope with large fluctuations in bandwidth requirement. Bandwidth smoothing techniques can reduce the burst of a variable-bit rate stream by prefetching data at a series of fixed rates and simplifying the allocation of resources in the video servers and the network. In this paper, the proposed smoothing algorithm results in the optimal transmission plans for (1) the smallest bandwidth requirements, (2) the minimum number of changes in transmission rate, and (3) the minimum amount of the server process overhead. The advantages of the proposed smoothing algorithm have been verified through the comparison with the existing smoothing algorithms in diverse environments.

• The KIPS Transactions:PartB
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• v.11B no.6
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• pp.625-632
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• 2004

Since the H.264 encoding adopts both forward prediction and hi-direction prediction modes and exploits Variable Length Coding(VLC), the amount of data generated from video encoder varies as Flaying time goes by. The fixed bit rate encoding system which has limited transmission channel capacity uses a buffer to control output bitstream It's necessary to control the bitstream to maintain within manageable range so as to protect buffer from overflow or underflow. With existing bit amount control algorithms, the $\lambda_{MODE}$ which is relationship between distortion value and quantization parameter often excesses normal value to end up with video error. This paper proposes an algorithm to protect buffer from overflow or underflow by introducing a new quantization parameter against distortion value of H.264 video data. The test results of 6 exemplary data show that the proposed algorithm has the same PSNR as and up to 8% reduced bit rate against existing algorithms.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.11a
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• pp.96-99
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• 2010

H.264/AVC 부호화 표준은 영상의 특성을 반영하기 위해 $16{\times}16$부터 $4{\times}4$ 크기의 가변적인 블록을 이용하여 부호화 효율을 높인다. 하지만 이로 인해 부호기의 복잡도가 증가된다. 부호기 복잡도를 증가시키는 여러 요인 중, H.264/AVC의 모드 결정은 부호기의 복잡도를 증가시키는 주요인이다. 본 논문에서는 IPPP구조에서 비트율 왜곡값을 이용하여 고속으로 매크로블록의 모드를 결정하는 방법을 제안한다. 인트라 화면에서의 인트라 $4{\times}4$, 인트라 $16{\times}16$의 비트율 왜곡 평균값으로 영상에 적응적인 최대 임계값과 최소 임계값을 결정한다. 다음, $16{\times}16$, $16{\times}8$, $8{\times}16$ 인터 모드의 비트율 왜곡값이 최대 임계값과 최소 임계값으로 분할한 범위 중 어느 곳에 해당하는지를 살펴보고, 이에 따라 인트라 모드 결정 단계를 선택적으로 결정한다. 제안하는 알고리즘은 기존의 H.264/AVC에 비해 부호화 효율의 큰 감소 없이 평균 27.42%의 부호화 시간을 감소시켰다.

• Journal of Broadcast Engineering
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• v.6 no.2
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• pp.148-159
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• 2001

This paper presents a selective duality control system to control relative picture quality among the video programs in terms of Peak Signal-to-Noise Ratio (PSNR) . The selective quality control system allows variable bit rate (VBR) for each video program to maintain the pre-determitted relative picture Quality among aggregated video programs while keeping a constant bit rate for alt programs to be transmitted over a single constant bit rate (CBR) channel. Thus is achieved by simultaneous controlling the video encoders to generate VBR video streams at the central controller. furthermore, we also suggest a buffer regulation method based on the analysis of the constraints Imposed by sender/receiver buffer sizes and the total transmission rate. Through various simulation results, it is found that the proposed quality control system guarantees that the video buffers neither overflow nor underflow and the quality control errors do not exceed 0.1 dB.

• Journal of the Korea Computer Industry Society
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• v.6 no.5
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• pp.801-812
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• 2005

H.264/AVC is the international video coding standard which has significant improvements of coding efficiency. ITU-T(International Telecommunication Union-Telecommunication standardization sector) and MPEG(Moving Picture Experts Group) adopts various complex coding tool such as variable block size motion, multiple reference frames, quarter-pel motion estimation/compensation (ME/MC) and rate-distortion(RD) optimization, etc. H.264 reference model employs complex mode decision technique based on RD optimization which requires high computational complexity. In this paper, we propose an efficient coding mode decision based on the cost distribution of RD in the macroblock coding mode sequence. Simulation results show that the proposed method reduces encoding time by 34% on average and save the number of computing RD cost by 82%.

• Journal of the Korean Institute of Illuminating and Electrical Installation Engineers
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• v.18 no.1
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• pp.129-136
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• 2004

In this paper, we proposed quad variable rates ADPCM coding method for efficient speech transmission and real time porcessing is implemented on TMS320C6711-DSP. The modified ADPCM with four variable coding rates, 16[kbps], 24[kbps], 32[kbps] and 40[kbps] are used for speech window samples for good quality speech transmission at a small data bits and real time encoding and decoding is implemented using DSP. ZCR is used to identify the influence of the noise on the speech signal and to decide the rate change threshold. For noise superior signals, low coding rates are applied to minimize data bit and for noise inferior signals, high coding rates are applied to enhance the speech quality. In most speech telecommunications, silent period takes more than half of the signals, speech quality close to 40[kbps] can be obtained at comparabley low data bits and this is shown by simulation and experiments. TMS320C6711-DSK board has 128K flash memory and performance of 1333MIPS and has meets the requirements for real time implementation of proposed coding algorithm.