• Journal of the Korea Society for Simulation
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• v.21 no.4
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• pp.47-56
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• 2012

In this study, we use a parallel simulator PASENS(Parallel SEnsor Network Simulator) to predict power consumption and data reception rate of the hierarchical routing protocols for sensor network - LEACH (Low-Energy Adaptive Clustering Hierarchy), TL-LEACH (Two Level Low-Energy Adaptive Clustering Hierarchy), M-LEACH (Multi hop Low-Energy Adaptive Clustering Hierarchy) and LEACH-C (LEACH-Centralized). According to simulation results, M-LEACH routing protocol shows the highest data reception rate for the wider area, since more sensor nodes are involved in the data transmission. And LEACH-C routing protocol, where the sink node considers the entire node's residual energy and location to determine the cluster head, results in the most efficient energy consumption and in the narrow area needed long life of sensor network.

• JSTS:Journal of Semiconductor Technology and Science
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• v.15 no.3
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• pp.404-416
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• 2015

Small-area, low-power coarse and fine frequency detectors (FDs) are proposed for an adaptive bandwidth referenceless CDR with a wide range of input data rate. The coarse FD implemented with two flip-flops eliminates harmonic locking as long as the initial frequency of the CDR is lower than the target frequency. The fine FD samples the incoming input data by using half-rate four phase clocks, while the conventional rotational FD samples the full-rate clock signal by the incoming input data. The fine FD uses only a half number of flip-flops compared to the rotational FD by sharing the sampling and retiming circuitry with PLL. The proposed CDR chip in a 65-nm CMOS process satisfies the jitter tolerance specifications of both USB 3.0 and USB 3.1. The proposed CDR works in the range of input data rate; 2 Gb/s ~ 8 Gb/s at 1.2 V, 4 Gb/s ~ 11 Gb/s at 1.5 V. It consumes 26 mW at 5 Gb/s and 1.2 V, and 41 mW at 10 Gb/s and 1.5 V. The measured phase noise was -97.76 dBc/Hz at the 1 MHz frequency offset from the center frequency of 2.5 GHz. The measured rms jitter was 5.0 ps at 5 Gb/s and 4.5 ps at 10 Gb/s.

• The Journal of the Acoustical Society of Korea
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• v.42 no.5
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• pp.422-428
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• 2023

Ultrafast ultrasound imaging has been applied to various imaging approaches, including shear wave elastography, ultrafast Doppler, and super-resolution imaging. However, these methods are still challenging in real-time implementation for three Dimension (3D) or portable applications because of their massive data rate required. In this paper, we proposed an adaptive quantization method that effectively reduces the data rate of large Radio Frequency (RF) data. In soft tissue, ultrasound backscatter signals require a high dynamic range, and thus typical quantization used in the current systems uses the quantization level of 10 bits to 14 bits. To alleviate the quantization level to expand the application of ultrafast ultrasound imaging, this study proposed a depth-sectional quantization approach that reduces the quantization errors. For quantitative evaluation, Field II simulations, phantom experiments, and in vivo imaging were conducted and CNR, spatial resolution, and SSIM values were compared with the proposed method and fixed quantization method. We demonstrated that our proposed method is capable of effectively reducing the quantization level down to 3-bit while minimizing the image quality degradation.

• The Transactions of the Korea Information Processing Society
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• v.7 no.2S
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• pp.649-655
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• 2000

Real-time application programs have constraints which need to be met between media-data. These constraints represents the delay time ad quality of service between media-data to be presented. In order to efficiently describe the delay time and quality of service, a new synchronization mechanism is needed. Proposed paper is a dynamic synchronization that minimized the effects of adaptive transmission delay time. That is, the method meets the requirements of synchronization between media-dat by handling dynamically the adaptive waiting time resulted from variations of delay time. In addition, the mechanism has interval adjustment using maximum delay jitter time. This paper decreases the data loss resulted from variation of delay time and from loss time of media-data by means of applying delay jitter in order to deal with synchronization interval adjustment. Plus, the mechanism adaptively manages the waiting time of smoothing buffer, which leads to minimize the gap from the variation of delay time. The proposed paper is suitable to the system which requires the guarantee of high quality of service and mechanism improves quality of services such as decrease of loss rate, increase of playout rate.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2009.01a
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• pp.668-672
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• 2009

In wireless ad-hoc network, knowing the available bandwidth of the time varying channel is imperative for live video streaming applications. This is because the available bandwidth is varying all the time and strictly limited against the large data size of video streaming. Additionally, adapting the encoding rate to the suitable bit-rate for the network, where an overlarge encoding rate induces congestion loss and playback delay, decreases the loss and delay. While some effective rate controlling methods have been proposed and simulated well like VTP (Video Transport Protocol) [1], implementing to cooperate with the encoder and tuning the parameters are still challenging works. In this paper, we show our result of the implementation experiment of VTP based encoding rate controlling method and then introduce some techniques of our parameter tuning for a video streaming application over wireless environment.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.7
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• pp.125-131
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• 1989

This paper proposes a new ABTC (Adaptive Block Truncation Coding) algorithm which improves the BTC algorithm for image data compression. A new adaptive block truncation coding which adopts a selective coding scheme depending on the local characteristics of an image has been described. The characteristics of the ABTC algorithm can be summarized as high compression ratio and the algorithm simplicity. Using this algorithm, color images can be coded at a variable bit rate from 1.0 (bit/pel) to 2.56 (bit/pel) and high compression rate (1.3-105 bit/pel) can be achieved without conspicuous image degradation compared with original images.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.11
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• pp.2068-2086
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• 2011

Based on the quincunx sub-sampling grid, the New Interleaved Hierarchical INTerpolation (NIHINT) method is recognized as a superior pyramid data structure for the lossless and progressive coding of natural images. In this paper, we propose a new image interpolation algorithm, Edge Adaptive Hierarchical INTerpolation (EAHINT), for a further reduction in the entropy of interpolation errors. We compute the local variance of the causal context to model the strength of a local edge around a target pixel and then apply three statistical decision rules to classify the local edge into a strong edge, a weak edge, or a medium edge. According to these local edge types, we apply an interpolation method to the target pixel using a one-directional interpolator for a strong edge, a multi-directional adaptive weighting interpolator for a medium edge, or a non-directional static weighting linear interpolator for a weak edge. Experimental results show that the proposed algorithm achieves a better compression bit rate than the NIHINT method for lossless image coding. It is shown that the compression bit rate is much better for images that are rich in directional edges and textures. Our algorithm also shows better rate-distortion performance and visual quality for progressive image transmission.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.12
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• pp.6017-6037
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• 2017

In a multi-resolution image encoding system, the image is encoded into a single file as a layer of bit streams, and then it is transmitted layer by layer progressively to reduce the transmission time across a low bandwidth connection. This encoding scheme is also suitable for multiple decoders, each with different capabilities ranging from a handheld device to a PC. In our previous work, we proposed an edge adaptive hierarchical interpolation algorithm for multi-resolution image coding system. In this paper, we enhanced its compression efficiency by adding three major components. First, its prediction accuracy is improved using context adaptive error modeling as a feedback. Second, the conditional probability of prediction errors is sharpened by removing the sign redundancy among local prediction errors by applying sign flipping. Third, the conditional probability is sharpened further by reducing the number of distinct error symbols using error remapping function. Experimental results on benchmark data sets reveal that the enhanced algorithm achieves a better compression bit rate than our previous algorithm and other algorithms. It is shown that compression bit rate is much better for images that are rich in directional edges and textures. The enhanced algorithm also shows better rate-distortion performance and visual quality at the intermediate stages of progressive image transmission.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.12 no.5
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• pp.810-816
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• 2001

This paper proposes a Truncated Type-II Hybrid ARQ scheme using an adaptive modulation system to achieve high throughput data transmission systems for mobile communication systems. In this paper, the adaptive modulation system analyzed in Nakagami (m-distribution) fading channel environment. The adaptive modulation system controls the modulation level and symbol rate according to the Nakagami fading parameter(m). When the received $E_bN_0$ is high or the Nakagami fading parameter m is high, the propose system selects higher modulation level and higher symbol rate to increase throughput. On the other hand, this system selects lower modulation level and lower symbol rate to prevent throughput performance degradation when the received $E_bN_0$ is low. The modulation method have been adopted QPSK(Quadrature Phase Shift Keying), 16 QAM(Quadrature Amplitude Modulation), 64 QAM, 256 QAM. Therefore, Adaptive Modulation Systems with Truncated Type-II Hybrid ARQ Scheme is proper for mobile and radio for mobile and radio data communication system that require high reliability and delay-limited applications.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.5 no.4
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• pp.661-667
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• 2001

This paper proposes a Truncated Type-II Hybrid ARQ scheme using an adaptive modulation system to achieve high throughput data transmission systems for DS-CDMA cellular mobile communication systems. In this paper, the adaptive modulation system analyzed in Nakagami (m-distribution) fading channel environment. The adaptive modulation system controls the modulation level and symbol rate according to the Nakagami fading parameter( m). When the received Eb/No is high or the Nakagami fading parameter m is high, the propose system selects higher modulation level and higher symbol rate to increase throughput. On the other hand, this system selects lower modulation level and lower symbol rate to prevent throughput performance degradation when the received Eb/No is low. The modulation method have been adopted QPSK 16QAM, 64QAM, 256QAM. Therefore, adaptive modulation systems with Truncated Type II Hybrid ARQ Scheme is proper for mobile and radio data communication system that require high reliability and delay-limited applications.