• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1487-1504
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• 1997

In this paper, the two-stage variable block-size multiresolution motion algorithm is proposed for an interframe coding scheme in the wavelet decomposition. An optimal bit allocagion between motion vectors and the prediction error in sense of minimizing the total bit rate is obtained by the proposed algorithm. The proposed algorithm consists of two stages for motion estimatation and only the first stage can be separated and run on its own. The first stage of the algorithm introduces a new method to give the lower bit rate of the displaced frame difference as well as a smooth motion field. In the second stage of the algorithm, the technique is introduced to have more accurate motion vectors in detailed areas, and to decrease the number of motion vectors in uniform areas. The algorithm aims at minimizin gthe total bit rate which is sum of the motion vectors and the displaced frame difference. The optimal bit allocation between motion vectors and displaced frame difference is accomplished by reducing the number of motion vectors in uniform areas and it is based on a botom-up construction of a quadtree. An entropy criterion aims at the control of merge operation. Simulation resuls show that the algorithm lends itself to the wavelet based image sequence coding and outperforms the conventional scheme by up to the maximum 0.28 bpp.

• 한국초전도학회:학술대회논문집
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• v.14
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• pp.64-64
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• 2004

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.10 no.11
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• pp.5381-5399
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• 2016

Currently, the quantize-and-forward (QF) scheme with high order modulation and quantization has rather high complexity and it is thus impractical, especially in multiple relay cooperative communications. To overcome these deficiencies, an improved low complex QF scheme is proposed by the combination of the low order binary phase shift keying (BPSK) modulation and the 1-bit and 2-bit quantization, respectively. In this scheme, the relay selection is optimized by the best relay position for best bit-error-rate (BER) performance, where the relays are located closely to the destination node. In addition, an optimal power allocation is also suggested on a total power constraint. Finally, the BER and the achievable rate of the low order 1-bit, 2-bit and 3-bit QF schemes are simulated and analyzed. Simulation results indicate that the 3-bit QF scheme has about 1.8~5 dB, 4.5~7.5 dB and 1~2.5 dB performance gains than those of the decode-and-forward (DF), the 1-bit and 2-bit QF schemes, at BER of $10^{-2}$, respectively. For the 2-bit QF, the scheme of the normalized Source-Relay (S-R) distance with 0.9 has about 5dB, 7.5dB, 9dB and 15dB gains than those of the distance with 0.7, 0.5, 0.3 and 0.1, respectively, at BER of $10^{-3}$. In addition, the proposed optimal power allocation saves about 2.5dB much more relay power on an average than that of the fixed power allocation. Therefore, the proposed QF scheme can obtain excellent features, such as good BER performance, low complexity and high power efficiency, which make it much pragmatic in the future cooperative communications.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 1995.06a
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• pp.109-112
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• 1995

A rate control strategy in the MPEG is critical to the quality of the reconstructed image sequence. In this paper, were propose a rate control algorithm which can improve the coding performance for a given bit constraint. The proposed algorithm use the distortion-rate curve for the fine adjustment of the quantization scaling factor of each region. This algorithm can also handle the problem due to scene change effectively. It can be easily applicable to existing MPEG coders. Simulation results show that the performance for the algorithm is better than the rate control algorithm in the MPEG-2 TM5.

• Proceedings of the IEEK Conference
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• 2000.06d
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• pp.98-101
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• 2000

On CELP type Vocoders G.723.1 6.3kbps/5.3kbps Dual Rate Speech Codec, which is developed for Internet Phone and videoconferencing, uses VAD(Voice Activity Detection)/CNG (Comfort Noise Generator) in order to reduce the bit rate in a silence period. In order to reduce the bit rate effectively in this paper, we first set the boundary condition of the energy threshold to prevent the consumption of unnecessary processing time, and use three decision rules to detect an active frame by energy, pitch gain and LSP distance. To evaluate the performance of the proposed algorithm we use silence-inserted speech data with 0, 5, 10, 20dB of SNR. As a result when SNR is over 5dB, the bit rate is reduced up to about 40% without speech degradation and the processing time is additionally decreased.

• The Journal of the Acoustical Society of Korea
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• v.24 no.2E
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• pp.61-66
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• 2005

Experimental work in underwater acoustic communications using passive phase conjugation has shown that the demodulation error depends on the relative drift rate between the source and receiver [Rouseff et al., IEEE J. Oceanic Eng. 26, 821-831 (2001)]. The observed effect involves the mismatch between the initial impulse response and the subsequent response after the source or receiver has changed locations. In the present work, the effect of drifting source is analyzed by numerical simulations and compared to the experimental results. The communications bit error rate is qualified as a function of drift rate, drifting direction, and source-receiver range.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.208-216
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• 2007

Wireless Sensor Networks (WSNs) are characterized by low capacity on each nodes and links. Wireless links have high bit error rate (BER) parameter that changes frequently due to the changes on network topology, interference, etc. To guarantee reliability in an error-prone environment, a retransmission mechanism can be used. In this mechanism, the number of retransmissions is used as a parameter that controls reliability requirement level. In this paper, we propose an Error Adaptive Transport Protocol (EATP) for WSNs that updates the number of retransmissions regularly to guarantee reliability during bit error rate changes as well as to utilize energy effectively. The said algorithm uses local information, thus, it does not create overhead problem.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12C
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• pp.1209-1216
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• 2003

The transcoding is needed to modify the bit rate of the video bitstream in accordance with the bandwidth of a variety of channels in order to provide users with a realtime multimedia service and more stable and credible VOD(Video On Demand) service on various wire/wireless network. This paper suggests how to guarantee the video quality in the transcoding process, taking the activity and direction of the macroblock into consideration, thereby reducing the calculation. The threshold of the macroblock's activity and direction is applicable to the different videos, with the DCT coefficients from the intermediate data in the decoding process. Moreover, the suggested video transcoding algorism enables us to maintain the proper video quality and reduce the calculation through the video bit rate change experiment in H.263 environment.

• Journal of Digital Convergence
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• v.10 no.5
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• pp.233-238
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• 2012

In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech waveform in case of exist a voiced and an unvoiced consonants in a frame. To solve this problem, this paper present a method of LMS-MPC uses individual pitch and LMS(Least Mean Square). I evaluate the MPC and LMS-MPC using LMS. As a result, SNRseg of LMS-MPC was improved 1.5dB for female voice and 1.3dB for male voice respectively. Compared to the MPC, SNRseg of LMS-MPC has been improved that I was able to control the distortion of the speech waveform finally. And so, I expect to be able to this method for cellular phone and smart phone using excitation source of low bit rate.

• 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.