• The Journal of the Acoustical Society of Korea
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• v.22 no.5
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• pp.418-424
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• 2003

In this paper we analyzed the performance of data transmission using BFSK modulation. The system performances were evaluated by the experiments in water tank. As a result we showed the influences of reverberation due to the multipath. In order to simplify the experiment procedure the channel coding etc. were omitted. The experimental result shows that the maximum transmission data rate in used water tank is about 800 bps. We also verified that the reverberation effect m reduced using a deconvolution with a measured channel impulse response. This method improved the bit rate by about 100 bps than simple noncoherent demodulator at bit error rate of 10/sup -3/.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.3 no.1 s.4
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• pp.109-116
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• 2004

The signal in wireless multi-path channel is affected by fading and ISI because of high data rate transmission, so the signal has the high error rate. The present modulation and demodulation method of DSRC system can not expect sufficient for providing data service over 1 Mbps, so the channel equalization and advanced modulation and demodulation methods are required. OFDM method is generally Inon as an effective technique for high data rate transmission system, since it can prevent ISI by inserting a guard interval. However, a guard interval longer than channel delay spread has to be used in each OFDM symbol period, thus resulting a considerable loss in the efficiency of channel utilization. Therefore the equalizer is necessary to cancel ISI to accommodate advanced ISI service with higher bit rate and longer channel delay spread condition. In this thesis, the channel equalizer for the OFDM-DSRC system was designed and its performance in a multi-path fading environment was evaluated with computer simulation. As a result, the performance of Pseudo LMMSE equalizer for the OFDM-DSRC has been improved comparing with LS equalizer at higher bit rate transmission system.

• Journal of KIISE:Information Networking
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• v.27 no.2
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• pp.206-218
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• 2000

Retransmissions on the DLC layer are essential to ABR service providing the low CLR (cell loss ratio) over the unreliable wireless channel with high bit error rate. In the wireless ATM, the DLC layer below ATM layer performs the retransmission and reordering of the cells to recover the cell loss over the wireless channel and by doing so, the effect of the wireless channel characteristics with high bit error rate can be minimized on the ATM layer which is designed under the assumption of the low bit error rate. We propose, in this paper, the schemes to reflect the changes of the transmission rate over the wireless channel on the ABR rate control. Proposed scheme can control the source rate to the changes of the transmission rate over the wireless channel and reduce the required buffer size in the AP (access point). In the simulation, we assume that the DLC layer can inform the ATM layer of the wireless channel quality as good or bad. Our simulation results show that the proposed schemes require the smaller buffer size compared with the existing scheme, enhanced dynamic max rate control algorithm (EDMRCA). It is also shown that the scheme with the intelligent DLC which adjusts the rate to the wireless channel quality not only provides the low CLR with smaller buffer requirement but also improves the throughput by utilizing the wireless bandwidth more efficiently.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.12 no.12
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• pp.5843-5848
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• 2011

To improve the efficiency of the channel spectrum and to reduce the power consumption of the system in EVRC, the voice signal is compressed and transmitted only when the user speaks to. In addition to this, voice frames are divided into three rates 1, 1/2 and 1/8 and each frame is handled differently. For example, we assumed that the input is silence region if the 1/8 rate is used. In this paper, the sections are firstly separated into the voiced speech signal region, unvoiced speech signal region, and silence region by using distribution characteristics of LSP parameters. Then the paper suggested to encode 1 rate for the voiced speech signal, 1/2 rate for the unvoiced speech signal region, 1/8 rate for the silence region. In other words, traditional way of transmission is used when sending full rate in the EVRC. However, when sending half rate, the voice is firstly distinguished between voiced and unvoiced. If the voice is distinguished as voiced, voice is converted into full rate before the transmission. If it is distinguished as silence, EVRC's basic rate is applied. In the experimental results with SNR, ASDM, transmission bit rate measurement, we have demonstrated that voice quality was improved by using the proposed algorithm.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.16 no.11
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• pp.1054-1062
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• 1991

In this paper a communication system using direct sequence spread spectrum (DS/SS) technique is constructed to transmit burst format data over power line channel with impulsive noise and narrowband interferences. Fast code synchronization is acquired by digital matched filter and data decision is accomplished by sampling pulses. In order to examine the performance of the power line communication system, but error rate and packet loss rate are measured over the simulation channel with various noise sources. When the packet composed of 1-bit preamble and 63-bit data is transmitted under very high burst impulsive noise, the bit error rate is about 10$^3$-10$^4$ and the packet loss rate is below 0.07.

• 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 Institute of Information and Communication Engineering
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• v.7 no.5
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• pp.887-892
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• 2003

In this paper, we propose a simple scaling algorithm for CDMA mobile communications where a voice traffic signals are transmitted by individual one of several data rates at every frames. The traditional method is based on using look-up table called SMT(symbol metric table), but the proposed algorithm is real-time direct scaling method through simple bit manipulations without lookup table. The bit error rate performance is calculated by computer simulation over AWGN and Rayleigh fading channels. From the results, it is shown that the proposed algorithm outperforms the traditional SMT method on Rayleigh channel by 0.3∼0.8dB, while achieving the less H/W complexity.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.4C
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• pp.200-206
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• 2005

Adaptive transmission techniques can improve the performance of wireless communication system by adaptively changing the transmission parameter such as modulation, code-rate, and power according to the channel state. For orthogonal frequency division multiplexing (OFDM) system, the adaptive transmission techniques can be applied to each subcarrier unit. In this paper, we consider the adaptive code-rate OFDM system in which optimal code-rate is applied to each subcarrier according to the subchannel state. Performance analysis show that $3\sim6$dB gain of SNR or up to $30\sim50\%$ increase of data rate are achieved in the condition of bit error rate $10^{-6}$.

• Journal of the Korean Institute of Telematics and Electronics
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• v.16 no.1
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• pp.14-21
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• 1979

This paper deals with tIne interface error in the on-line computer systems by using the FSK radio communications. The wideband frequency shrift keying method is used for briary data transmission between the remote terminals and the main computer. To mintmize the error rate In the decoder systems of the main computer, a synchronizing pulse is added to the frame, so that the phase In both receiver and transmitter are synchronized. When the information signal with a constant error bit is received through FSK, It is designed to use the microprocessor for calculation of error bit. As results, most bit error are caused in FSK radio communications. and a few error bit Is me - asured to enter the mirroprocessor from the input buffer.