• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.7 no.5
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• pp.418-429
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• 1996

The number of bits per message and the number of tones in the frequency-hopping sequence are determined by the available bandwidth and the data rate of each user. These parameters in turn determine the tone duration which strongly influences the vulnerability of the system to transmission distortions. In this paper, an address code which is assigned to each individual user was employed in order to reduce the collisions or hit. Also the frequency band is divided into several subbands and each user transmits multitone frequency per subband per chip. And the new system which is to increase the duration of each tone by increasing the total number of system frequencies that has been proposed. It is found that an optimum value bit, tone, number of frequencies per chirp can improve the err performance. This flexibility slightly increases maximum efficiecy and makes the the system less vulnerable to multipath delay. So, It is found that as the nuber of user increased 50%, the efficiency as a tuncion of the bandwidth to user'rate ratio improve 20%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.1A
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• pp.50-58
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• 2008

The use of space-division multiple access(SDMA) in the downlink of multiuser multi-input/multi-output(MIMO) wireless transmission systems can provide substantial gains in system throughput. When the channel state information(CSI) is available at the transmitter, a considerable performance improvement can be attained by adapting the transmission rates to the reported CSI. In addition, to combat frequency selective fadings in wideband wireless channels, bit-interleaved coded OFDM(BIC-OFDM) modulation schemes are employed to provide reliable packet delivery by utilizing frequency diversity through channel coding. In this paper, we propose an adaptive modulation and coding(AMC) scheme combined with an opportunistic scheduling technique for the MIMO BIC-OFDM with bandwidth-limited feedback channels. The proposed scheme enhances the link performance by exploiting both the frequency diversity and the multiuser diversity. To reduce the feedback information, the proposed AMC scheme employs rate adaptation methods based on an OFDM symbol rather than on the whole subchannels. Simulation results show that the proposed scheme exhibits a substantial performance gain with a reasonable complexity over single antenna systems.

• The Transactions of the Korea Information Processing Society
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• v.5 no.6
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• pp.1610-1619
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• 1998

The ABR service class provides feedback based traffic control to transport bursty data traffic efficiently. Feedback based congestion control has first been studied to be applied to unicast connections. Recently. several congestion control algorithms for multicast connections have also been proposed as the number of ABR applications requiring multicast increases. With feedback based congestion control, the effectiveness of a traffic control scheme diminishes as propagation delay increases. Especially for a multicast connection, a remote destination may suffer unfair service compared to a local destination due to the delayed feedback. Amelioration of the disadvantages caused by feedback delay is therefore more important for remote destinations in multicast connections. This paper proposes a new switch behavior to provide effective feedback based mathc control for rentoh destinations. The proposed switches adjust the service rate dynamically in accordance woth the state of the downstream, that is, the congestion of the destinaion is immediately controlled by the nearest apstream switch before the source to ramp down the transmission rate of the connection. The proposed switch has an implementation overhead to have a separate buffer for each VC to adjust the service rate in accordance with a backward Rm cell of each VC. The buffer requirement id also increased at intermediate switches. Simulation results show that the proposed switch reduces the cell loss rate in both the local and the remote destinations and slso amelioratd the between the two destinations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.12A
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• pp.991-997
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• 2009

Transmit beamforming is simple method to achieve the full diversity gain that is available in multiple antenna(MIMO) wireless systems. Unfortunately, the prior condition to achieve this gain requires perfect channel knowledge at both transmitter and receiver, which is impractical on account of limited feedback link. Therefore, for the practical system, codebook based feedback scheme is often employed, where the beamforming vector is selected from the codebook to maximize the output signal-to-noise ratio (SNR) at receiver, and the receiver only sends back the index of the best beamforming vector to the transmitter. In this paper we derive analytical expression of average bit error rate (BER) for the codebook based transmit beamforming MISO system over the feedback error channel. Using this analytical result, we present optimum codebook indexing scheme to improve the performance of this system. From some selected numerical examples we show that our proposed codebook indexing scheme can provide nonnegligible performance improvements in terms of average BER over the severe feedback error channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.36 no.11C
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• pp.673-681
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• 2011

In this paper, we design multiple antenna relay transmission schemes of low complexity to enhance the spatial diversity in orthogonal frequency division multiplexing (OFDM) systems. The relay scheme underlined, can provide space time block coding (STBC) of OFDM signals in the time domain without IFFT and FFT operations with much reduced complexity. In this paper, we modify the conventional low-complexity STBC-OFDM relaying scheme to be compatible to the existing OFDM systems. In addition, we extend the proposed scheme for multiple antenna relays and provide performance enhancement strategies according to the channel quality information available at the relay. The proposed scheme is shown to improve the diversity and thereby to reduce the outage probability and coded bit error rate. Therefore, the proposed scheme will be promising for service quality improvement or coverage extension based on OFDM like wireless LANs and maritime communications.

• IEIE Transactions on Smart Processing and Computing
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• v.4 no.5
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• pp.371-378
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• 2015

In this paper, a different type of pulse width modulation (PWM) control scheme for a buck converter is introduced. The proposed buck converter uses PWM with frequency hopping and a low quiescent.current low dropout (LDO) voltage regulator with a power supply rejection ratio enhancer to reduce high spurs, harmonics and output voltage ripples. The low quiescent.current LDO voltage regulator is not described in this paper. A three-bit binary-to-thermometer decoder scheme and voltage ripple controller (VRC) is implemented to achieve low voltage ripple less than 3mV to increase the efficiency of the buck converter. An internal clock that is synchronized to the internal switching frequency is used to set the hopping rate. A center frequency of 2.5MHz was chosen because of the bluetooth low energy (BLE) application. This proposed DC-DC buck converter is available for low-current noise-sensitive loads such as BLE and radio frequency loads in portable communications devices. Thus, a high-efficiency and low-voltage ripple is required. This results in a less than 2% drop in the regulator's efficiency, and a less than 3mV voltage ripple, with -26 dBm peak spur reduction operating in the buck converter.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.2
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• pp.19-25
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• 2008

When there are numerous wireless communication systems co-existing in the limited available frequency resource, an unexpected time delay can be caused during the system switching. So, in order to reduce this time delay, a mode selection method is required. In this paper, we propose a mode selection method to minimize the time delay for multi-system wireless communication systems. For the sake of efficiency, the mode selection method is designed by analyzing the preamble characteristics of different standards. Instead of performing a full search, we propose the preamble partial search to reduce the time delay to a minimum. Simulated with Matlab in an additive white Gaussian noise(AWGN) environment with a signal to noise ratio(SNR) of 10dB and bit error rate(BER) of $10^{-6}$, we evaluated and showed the performance improvement gained by using our proposed mode selection method.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.9 no.4
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• pp.99-107
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• 2009

Visible Light Communication (VLC) is a promising wireless communication technology. It offers huge, worldwide available and free bandwidth without electro-magnetic interference, which makes it very attractive for RF-sensitive operating environments. We propose colored LED-based VLC system for hospital use which includes voice recognition system for operating the medical equipments. New Mr hlation Scheme based on the Light Color Space is suggested to overcome the effect of noise generated by background light. Different color space constellations for different symbol sizes are also suggested which would give better bit error rate performance. Finally, Slotted ALOHA or TDMA medium access control protocols are suggested for multi-user operations.

• The Journal of the Acoustical Society of Korea
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• v.21 no.6
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• pp.566-575
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• 2002

Low-titrate audio coding enables a number of Internet and mobile multimedia streaming service more efficiently. For the help of next-generation mobile telephone technologies and digital audio/video compression algorithm, we can enjoy the real-time multimedia contents on our mobile devices (cellular phone, PDA notebook, etc). But the limited available bandwidth of mobile communication network prohibits transmitting high-qualify AV contents. In addition, most bandwidth is assigned to transmit video contents. In this paper, we design a novel and simple method for reproducing high frequency components. The spectrum of high frequency components, which are lost by down-sampling, are modeled by the energy rate with low frequency band in Bark scale, and these values are multiplexed with conventional coded bitstream. At the decoder side, the high frequency components are reconstructed by duplicating with low frequency band spectrum at a rate of decoded energy rates. As a result of segmental SNR and MOS test, we convinced that our proposed method enhances the subjective sound quality only 10%∼20% additional bits. In addition, this proposed method can apply all kinds of frequency domain audio compression algorithms, such as MPEG-1/2, AAC, AC-3, and etc.

• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.