• 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.

• Journal of Internet Computing and Services
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• v.21 no.4
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• pp.35-41
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• 2020

This paper proposes an algorithm to improve the compression performance of the adaptive multi-rate (AMR) speech coding using the context tree weighting (CTW) method. AMR is the voice encoding standard adopted by IMT-2000, and supports 8 transmission rates from 4.75 kbit/s to 12.2 kbit/s to cope with changes in the channel condition. CTW as a kind of the arithmetic coding, uses a variable-order Markov model. Considering that CTW operates bit by bit, we propose an algorithm that re-orders AMR data and compresses them with CTW. To verify the validity of the proposed algorithm, an experiment is conducted to compare the proposed algorithm with existing compression methods including ZIP in terms of compression ratio. Experimental results indicate that the average additional compression rate in AMR data is about 3.21% with ZIP and about 9.10% with the proposed algorithm. Thus our algorithm improves the compression performance of AMR data by about 5.89%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.9
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• pp.817-825
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• 2013

The UHDTV system, which provides realistic service with ultra-high definite video and multi-channel audio, has been studied as a next generation broadcasting service. Since the conventional digital terrestrial transmission system is not capable to cover the increased transmission data rate of the UHDTV service, there are great necessity of researches about increase of data rate. Accordingly, the researches has been studied to increase the transmission data rate of the DVB-T2 system using dual-polarized MIMO technique and high order modulation. In order to optimize the MIMO DVB-T2 system where irregular LDPC codes are used, it is necessary to study the design of the bit mapper that matches the LDPC code and QAM symbols in MIMO channel. However, the research related to the design of the bit mapper has been limited to the SISO system. Therefore, this paper defines a new parameter that indicates the VND distribution of MIMO DVB-T2 system and performs the performance analysis according to the parameter which will be helpful for designing a MIMO bit mapper.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.97-100
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• 1999

Mobile network applications are growing and this requires a fast and efficient transport method between the BS(Base Station) and the MSC(Mobile Switching Center). One possible solution is to use ATM and a voice CODEC standard which compresses 64kbps voice data to less than 8kbps. The low bit tate and small-sized packets made by the voice CODEC imply that significant amount of link bandwidth would be wasted, if this small-sized packet is carried by one ATM cell. The cell assembly delay increases if one ATM cell is fully filled with the small-sized packets. For the bandwidth-efficient transmission of low-rate, short, and variable length packets in delay sensitive applications, AAL-2 was standardized. This paper evaluates performance of AAL-2 by using voice CODEC standard.

• Journal of the Korea Society of Computer and Information
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• v.17 no.5
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• pp.71-80
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• 2012

In this paper, applications that require low-speed data rate on personal area network to operate the Land Warrior system is proposed. These applications can refer to WSN(wireless sensor network) technology. However, this technology is not suitable to support various data transmission rate. A suitable CSMA/CA algorithm for Land Warrior System in order to solve these existing system problems is proposed. The proposed algorithm is designed to be variable CSMA/CA algorithm parameter, depending on data rate. For the evaluation of Land Warrior system model and CSMA/CA algorithm, we used Castalia. As a result of the simulation, it is found that the proposed system model can not only relieve loads of data processing, but also probability of collision was decreased.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.12C
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• pp.1192-1199
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• 2003

In this paper, we investigate the effect of pulse repetition frequency (PRF) and slot interval on the throughput performance of the ultra wide band (UWB) wireless communication system in multi-path channels, and based on these observations, a data throughput control using PRF and slot interval is proposed for maximizing the effective throughput. Recently, due to many desirable features of the UWB system, it has drawn much attention especially for short-range high-speed data transmission. The UWB system has two parameters to determine its data throughput; pulse repetition frequency and slot interval. In the multi-path channel with additive white Gaussian noise, the UWB system suffers from the inter-pulse interference (IPI) and noise, which result in degradation of system performance. The UWB system can vary the two parameters to maintain and/or improve the system performance. In this paper, we demonstrate the effects of the two parameters on the data throughput of the UWB system in various multi-path indoor channels through computer simulation, and show that the variable data rate approach designed based on the observations is superior to the fixed data rate one in terms of effective throughput performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.19 no.9
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• pp.1730-1738
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• 1994

Broadband code division multiple access(B-CDMA)previously proposed for personal communication services(PCS) is not suitable for high-speed data services with large variations in speed, because it is designed only for low-speed voice calls with small variations in speed. Meanwhile, hybrid intra-cell TDMA/inter-cell CDMA(Hybrid TDMA/CDMA) recently proposed could accomodate high-speed services easily because of its internal use of the TDMA scheme, but there would be much inefficiency in treating variable bit rate services due to the intrinsic properties of TDMA transmission scheme, In this paper, a media access control(MAC) protocol, called Resevation TDMA/CDMA, which uses a reservation scheme similar to the one in the packet reservation multiple access(PRMA) protocol to compensate for these shortcomings is proposed. In addition, the performance of this protocol is analyzed and compared with conventional protocols by computer simulation. According to the simulation results, it has been shown that the proposed Reservation TDMA/CDMA scheme could support various transmission rates of user data traffics easily, and it could utilize the variations of the treansmission rates in each active call more efficiently.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.6C
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• pp.615-623
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• 2003

This paper presents effective quality-of-service renegotiating schemes for streaming video. The conventional network supporting quality-of-service generally allows a negotiation at call setup. However, it is not efficient for the video application since the compressed video traffic is statistically non-stationary. Thus, we consider the network supporting quality-of-service renegotiations during the data transmission, and study effective quality-of-service renegotiating schemes for streaming video. Simple token bucket model, whose parameters are token filling rate and token bucket size, is adopted for the video traffic model. The renegotiating time instants and the parameters are determined by analyzing the statistical information of compressed video traffic. In this paper, two renegotiating approaches, i.e. fixed renegotiating interval case and variable renegotiating interval case, are examined. Finally, the experimental results are provided to show the performance of the proposed schemes.

• Asia Pacific Journal of Business Review
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• v.1 no.2
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• pp.1-16
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• 2017

The fundamental goal of the research was to verify if the Twin Deficits Hypothesis holds for the economy of Zambia using time series data from 1980-2014. The current account and budget deficit were employed as key variables. The exchange rate was also used as a transmission mechanism to see how it contributes in the nexus. Cointegration tests confirmed a long run association of the variables. After fitting the VECM model, Granger causality tests confirmed the existence of twin deficits for Zambia. The results supported uni-directional reverse causality. The exchange rate was shown to be more significant in the long run than in the short run. The implosion of the time series as shown by the predicted cointegration equation implies that unless drastic measures are taken to cure the deficits, using the current account as the major target variable, twin deficits will persist for some time. The major policy implication of this research is that given that Zambia is a primary commodity-dependent developing country subsisting largely on copper revenues to sustain the economy, there is a need to move away from "copper addiction," given the recent volatility of earnings of primary commodities (e.g. through diversification of the economy, import substitution, and other strategies).

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.4
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• pp.803-812
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• 2018

The Morse code was firstly used for a wire telegraph service on May 1884 in USA. After about 10 years Korean researchers developed the Morse code in 1884 and the Morse code was firstly introduced in Korea for the wire telegraph service in September 1885. For about 130 years later, the Hangul Jamo Morse code has been used without any serious modification up to now. The Morse code is the variable length code system which each codes have a different code length. The related reference shows that English alphabet Morse code was applied in the coding theory. However, a result of analysis showed that the Hangul Jamo Morse code was not applied in the coding theory as the english alphabet code system. This paper presents the analysis result of Hangul Jamo Morse code according to the variable length coding theory. As a result of the analysis, when Hangul Jamo are coded with application of the variable length coding theory, the transmission data rate could be increased up to 25% compare to the existing code system. This paper shows a new Hangul Jamo Morse code system based on the analysis result.