• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.1
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• pp.132-139
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• 2016

This study was carried out near the waters of Jango port, Dangjin-gun, Chungcheongnam-do by utilizing 10kHz carrier frequency, for the purpose of measurement and analysis of underwater channel environment of the Western sea. For the measurement of horizontal channel environment, the separation distance between transmitter and receiver is made differently in the range between 10m and 4000m. Meanwhile, for the measurement of vertical channel environment, transmission and receiving side ships are fixed as contacted each other and measured differently depending on their depth of submergence. In this study, the Coherence Bandwidth and the Coherence Time were estimated by analyzing the Power delay profile of the real sea based on the measured data, and analyzing the doppler frequency through frequency conversion of received tone-signal, respectively. This study is expected to become a base study in carrying out the frame design for underwater communication to improve the communication and secure the reliability of communication in future underwater channel environment.

• Journal of Internet Computing and Services
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• v.15 no.2
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• pp.19-31
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• 2014

In this paper, a parallel backoff delay procedure on multiple IEEE 802.15.4 channels and a channel searching method considering the frequency spectrum of WiFi traffic are studied for IEEE 802.15.4 nodes to avoid the interference from WiFi traffic. In order to search the channels being occupied by WiFi traffic, we analyzed the methods measuring the powers of adjacent channels simultaneously, checking the duration of measured power levels greater than a threshold, and finding the same periodicity of sampled RSSI data as the beacon frame by signal processing. In an wireless channel overlapped with IEEE 802.11 network, the operation of CSMA-CA algorithm for IEEE 802.15.4 nodes is explained. A method to execute a parallel backoff procedure on multiples IEEE 802.15.4 channels by an IEEE 802.15.4 device is proposed with the description of its algorithm. When we analyze the data measured by the experimental system implemented with the proposed method, it is observed that medium access delay times increase at the same time in the associated IEEE 802.15.4 channels that are adjacent each other during the generation of WiFi traffic. A channel evaluation function to decide the interference from other traffic on an IEEE 802.15.4 channel is defined. A channel searching method considering the channel evaluations on the adjacent channels together is proposed in order to search the IEEE 802.15.4 channels interfered by WiFi, and the experimental results show that it correctly finds the channels interfered by WiFi traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.6A
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• pp.533-539
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• 2002

It is well known the fact that turbo lodes has better performance as the number of iteration and the interleaver size increases in the AWGN channel environment. However, as the number of iteration and the interleaver size are increased, it is required much delay and computation for iterative decoding, and caused the flattening effect phenomenon which is very litter BER performance improvement at the arbitrary SNR. In this paper, We proposed the new asymmetric turbo codes, which consist of parallel concatenated turbo codes that use mixed types of component codes with different not only constraint length but also generate polynomial and analyzed its BER performance for log-MAP decoding algorithm with frame size of 128, 256, 512 and 1024 bits, and coding rate of 1/3. As a results of simulation, proposed asymmetric turbo codes verify that its BER performance is superior to conventional symmetric turbo codes. It can be also observed that the flattening effect phenomenon is very reduced by applying the proposed asymmetric turbo codes. It gains respectively 1.7dB ~2.5dB and 2.0dB~2.5dB SNR improvements in the case of short frame(128, 256) and large frame(512, 1024) size for the BER $10_{-4}$>/TEX> region.

• Journal of the Korea Society of Computer and Information
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• v.18 no.11
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• pp.77-86
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• 2013

In this paper, we propose a fast video retargeting method which preserves the contents of a video and converts the image size. Since the conventional Seam Carving which is the well-known content-aware image retargeting technique uses the dynamic programming method, the repetitive update procedure of the accumulation energy is absolutely needed to obtain seam. The energy update procedure cannot avoid the processing time delay because of many operations by the image full-searching. By applying the proposed method, frames which have similar features in video are classified into a scene, and the first frame of a scene is resized by the modified Seam Carving where multiple seams are extracted from candidate seams to reduce the repetitive update procedure. After resizing the first frame of a scene, all continuous frames of the same scene are resized with reference to the seam information stored in the previous frame without the calculation of the accumulation energy. Therefore, although the fast processing is possible with reducing complexity and without analyzing all frames of scene, the quality of an image can be analogously maintained with an existing method. The experimental results show that the proposed method can preserve the contents of an image and can be practically applied to retarget the image on real time.

• ETRI Journal
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• v.36 no.5
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• pp.721-729
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• 2014

We propose a novel phase-based method for single-channel speech enhancement to extract and enhance the desired signals in noisy environments by utilizing the phase information. In the method, a phase-dependent a priori signal-to-noise ratio (SNR) is estimated in the log-mel spectral domain to utilize both the magnitude and phase information of input speech signals. The phase-dependent estimator is incorporated into the conventional magnitude-based decision-directed approach that recursively computes the a priori SNR from noisy speech. Additionally, we reduce the performance degradation owing to the one-frame delay of the estimated phase-dependent a priori SNR by using a minimum mean square error (MMSE)-based and maximum a posteriori (MAP)-based estimator. In our speech enhancement experiments, the proposed phase-dependent a priori SNR estimator is shown to improve the output SNR by 2.6 dB for both the MMSE-based and MAP-based estimator cases as compared to a conventional magnitude-based estimator.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.1
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• pp.38-40
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• 2014

In this paper, proposed method achieves delay reduction and throughput improvement by utilizing token-ring method in infrastructure network. Access Point gives token passing effect by successively transmitting ACK frame including a node's Association ID. Not only can this method considerably reduce time for medium access, but also improve throughput. Furthermore, AP offers more frequent medium access opportunity to node having highest data queue among nodes associated by AP. these method can evenly offer medium access opportunity according to Queue's volumes.

• Proceedings of the IEEK Conference
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• 2004.08c
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• pp.528-531
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• 2004

This paper presents the design and implementation of a QoS based IrDA in mobile phones. The IrDA standards have arised from the need to connect various mobile devices together. But currently, we have a difficult to connecting various mobile devices together. The difficult thing is to adjust the minimum turn around time. The minimum turn around time in IrDA deals with the time needed for a receiver circuit to recover following saturation by transmissions from the same device (turn around latency). This parameter corresponds to the required time delay between the last byte of the last frames sent by a station and the point at which it is ready the first byte of a frame from another station. This parameter comes into play when the link is turned around and is negotiated independency for each station. We solve this matter by using the Dynamic Adjustment Algorithm (DAA). The dynamic adjustment algorithm adjusts the minimum turn around time according to the numbers of devices. We apply our direct adjustment algorithm to our QoS based IrDA in mobile phones and show the improved performance of connection with various mobile phones together.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.10
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• pp.89-94
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• 2003

Passive Optical Network(PON) architecture is a very simple. However media access protocol is necessary to the PON because the ONUs should share a channel in the upsteam direction. We propose the design of dynamic bandwidth allocation algorithm for improved bandwidth utilization in the broad bandwidth access network that uses Ethernet PON. The object of our study shall be to decrease a unused remainder bandwidth of granted timeslot for a upstream traffic transmission. We propose that OLT notifies ONU of maximum transmission window during discovery and ONU request a amount under maximum transmission window after it checks a frame length. We confirm that the proposed algorithm shown improving performance from bandwidth utilization and queueing delay point of view through the simulation result.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.7 no.2
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• pp.366-385
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• 2013

In this paper, we propose a compressed sensing (CS) based Wyner-Ziv (WZ) codec using motion-aligned auto regressive model (MAAR) based side information (SI) extrapolation to improve the compression performance of low-delay distributed video coding (DVC). In the CS based WZ codec, the WZ frame is divided into small blocks and CS measurements of each block are acquired at the encoder, and a specific CS reconstruction algorithm is proposed to correct errors in the SI using CS measurements at the decoder. In order to generate high quality SI, a MAAR model is introduced to improve the inaccurate motion field in auto regressive (AR) model, and the Tikhonov regularization on MAAR coefficients and overlapped block based interpolation are performed to reduce block effects and errors from over-fitting. Simulation experiments show that our proposed CS based WZ codec associated with MAAR based SI generation achieves better results compared to other SI extrapolation methods.

• Journal of KIISE
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• v.43 no.3
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• pp.327-335
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• 2016

In this paper, we propose a buffer management scheme suitable for interactive multimedia services. We consider a typical delay optimization environment so that receiver buffer lengths vary according to the round trip time estimation. In this environment, we propose an optimization technique for minimizing the loss of information that may occur when a reduced buffer length forces I/P/B frames in the buffer to drop. We modeled our problem as a Knapsack Problem for which we used dynamic programing in order to find an approximate solution. The proposed technique is compared with the existing buffer management techniques. Through simulation studies, we found that our approach could increase PSNR, which is important to video quality.