• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9A
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• pp.694-701
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• 2003

In this paper, we present a hardware structure and new features of a smart antenna BTS (Base Transceiver Station) for CDMA2000 1X system. The proposed smart antenna BTS is a composite system consisting of many subsystems, i.e., array antenna element, frequency up/down converters, AD (Analog-to-Digital) and DA (Digital-to-Analog) converters, spreading/despreading units, convolutional encoder/Viterbi decoder, searcher, tracker, beamformer, calibration unit etc. Through the experimental tests, we found that the desired beam-pattern in both uplink and downlink communications is provided through the calibration procedure. Also it has been confirmed that the adaptive beamforming algorithm adopted to our smart antenna BTS is fast and accurate enough to support 4 fingers to each user. In our experiments, commercial mobile terminals operating PCS (Personal Communication System) band have been used. It has been confirmed that the smart antenna BTS tremendously improves the FER (Frame Error Rate) performance compared to the conventional 2-antenna diversity system.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.6 no.2
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• pp.175-180
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• 2002

In this paper, the system performace with the convolution rode using a Viterbi decoding and the one tap LMS(Least Meam Square) equalizer applied to the OFDM(Orthogonal Frequency Division Multiplexing) system, is analyzed through computer simulation. DMRS(Digital Microwave Radio System)is modeled as Rummler fading channel. In Simulation result, we known that the coding system improved about 3.6dB~10.5dB when BER is 10 $^3$and b is 0.1~0.2 in case of 16QAM(Qurdrature Amplitude Modulation). Also, we known that was improved about 19.7dB when the b is 0.1 and was demanded about 10.5dB when the b is 0.2 in case of 64QAM. we known that the soft decision improved about 2~0.9dB when the b is 0.1~0.2 in case of 16QAM and about 3.3~7.8dB in case of 64QAM. In the equalizer system, efficiency improved from the case of that Eb/No is more than 13dB.

• Journal of IKEEE
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• v.1 no.1 s.1
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• pp.138-155
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• 1997

Wireless optical communication is able to obtain high antenna gain as well as utilize to transmit the high-speed information with large capacity than the RF communication. However, the propagation path (atmosphere) is considered as an attenuator occured turbulence, absorption and scattering. These undesired phenomena diminish the amount of light that is collected at the receiver. To evaluate the effect of the atmospheric turbulance and scattering, this paper perform the ground-to-ground wireless optical LAN experiment by using the (2,1,6) convolutional coder and Viterbi decoder, and analyze numerically the earth-station antenna diameter due to the propagation path condition and upstream/downstream link.

• The Journal of the Acoustical Society of Korea
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• v.20 no.3
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• pp.63-69
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• 2001

A corpus-based lip sync algorithm for synthesizing natural face animation is proposed in this paper. To get the lip parameters, some marks were attached some marks to the speaker's face, and the marks' positions were extracted with some Image processing methods. Also, the spoken utterances were labeled with HTK and prosodic information (duration, pitch and intensity) were analyzed. An audio-visual corpus was constructed by combining the speech and image information. The basic unit used in our approach is syllable unit. Based on this Audio-visual corpus, lip information represented by mark's positions was synthesized. That is. the best syllable units are selected from the audio-visual corpus and each visual information of selected syllable units are concatenated. There are two processes to obtain the best units. One is to select the N-best candidates for each syllable. The other is to select the best smooth unit sequences, which is done by Viterbi decoding algorithm. For these process, the two distance proposed between syllable units. They are a phonetic environment distance measure and a prosody distance measure. Computer simulation results showed that our proposed algorithm had good performances. Especially, it was shown that pitch and intensity information is also important as like duration information in lip sync.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.37 no.9
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• pp.1-10
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• 2000

This paper proposes symbol decoding schemes combined with channel estimation techniques for coded orthogonal frequency division multiplexing (COFDM) systems in fading channels. sThe proposed symbol decoding schemes are consisted of a symbol decoding technique and channel estimation techniques. The symbol decoding based on Viterbi algorithm is achieved by matching the length of branch word from encoder trellis to the codeword length of symbol candidate on decoder trellis. Three combination schemes are described and their error performances are compared. The first scheme is to combine a symbol decoding technique with a training channel estimation technique. The second scheme joins a decision directed channel estimation technique to the first scheme. The time varying channel transfer functions are tracked by the decision directed channel estimation technique and the channel transfer functions used in the symbol decoder are updated every COFDM symbol. Finally, In order to reduce the effect of additive white Gaussian noise (AWGN) between adjacent subchannels, deinterleaved average channel estimation technique is combined. The error performances of the three schemes are significantly improved being compared with that of zero forcing equalizing schemes.

• Journal of Korea Multimedia Society
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• v.12 no.3
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• pp.329-334
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• 2009

When an optical disc is being played. the pick-up converts light to analog signal at first. The analog signal is equalized for removing the inter-symbol interference and then the equalized analog signal is converted into the digital signal for extracting the synchronized data and clock signals. There are a lot of algorithms that minimize the BER in extracting the synchronized data and clock when high. density optical disc like BD is being played in low speed. But if the high-density optical disc is played in high speed, it is difficult to adopt the same extraction algorithm to data PLL and PRML architecture used in low speed application. It is because the signal with more than 800MHz should be processed in those architectures. Generally, in the 0.13-${\mu}m$ CMOS technology, it is necessary to have the high speed analog cores and lots of efforts to layout. In this paper, the parallel data PLL and PRML architecture, which enable to process in BD 8x speed of the maximum speed of the high-density optical disc as the extracting data and clock circuit, is proposed. Test results show that the proposed architecture is well operated without processing error at BD 8x speed.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.1
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• pp.141-150
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• 2010

In this paper, performance of a turbo-coded OFDM system is analyzed and simulated in a power line communication channel. Since the power line communication system typically operates in a hostile environment, turbo code has been employed to enhance reliability of transmitted data. The performance is evaluated in terms of bit error probability. As turbo decoding algorithms, MAP (maximum a posteriori), Max-Log-MAP, and SOVA (soft decision viterbi output) algorithms are chosen and their performances are compared. From simulation results, it is demonstrated that Max-Log-MAP algorithm is promising in terms of performance and complexity. It is shown that performance is improved 3dB by increasing the number of iterations, 2 to 8, and interleaver length of a turbo encoder, 100 to 5000. The results in this paper can be applied to OFDM-based high-speed power line communication systems.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37 no.7A
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• pp.512-521
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• 2012

In this paper, we propose feasible combining rules for a decoding scheme of product codes to apply finite soft decision. Since the decoding scheme of product codes are based on complex tanh calculation with infinite soft decision, it requires high decoding complexity and is hard to practically implement. Thus, simple methods to construct look-up tables for finite soft decision are derived by analyzing the operations of the scheme. Moreover, we focus on using convolutional codes, which is popular for easy application of finite soft decision, as the horizontal codes of product codes so that the proposed decoding scheme can be properly implemented. Numerical results show that the performance of the product codes with convolutional codes using 4-bit soft decision approaches to that of same codes using infinite soft decision.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.8C
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• pp.806-813
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• 2009

This paper presents a modem chip design for high-speed wireless communications. Among the high-speed communication technologies, we design the UWB (Ultra-Wideband) modem SoC (System-on-Chip) Chip based on a MB-OFDM scheme which uses wide frequency band and gives low frequency interference to other communication services. The baseband system of the modem SoC chip is designed according to the standard document published by WiMedia. The SoC chip consists of FFT/IFFT (Fast Fourier Transform/Inverse Fast Fourier Transform), transmitter, receiver, symbol synchronizer, frequency offset estimator, Viterbi decoder, and other receiving parts. The chip is designed using 90nm CMOS (Complementary Metal-Oxide-Semiconductor) procedure. The chip size is about 5mm x 5mm and was fab-out in July 20th, 2009.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.9C
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• pp.720-724
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• 2008

Channel estimation in OFDM systems is usually carried out in frequency domain based on the least-squares (LS) method and the minimum mean-square error (MMSE) method with known pilot symbols. The LS estimator has a merit of low complexity but may suffer from the noise because it does not consider any noise effect in obtaining its solution. To enhance the noise immunity of the LS estimator, we consider estimation noise in time domain. Residual noise existing at the estimated channel coefficients in time domain could be reduced by reasonable selection of a threshold value. To achieve this, we propose a channel-estimation method based on a time-domain threshold which is a standard deviation of noise obtained by wavelet decomposition. Computer simulation shows that the estimation performance of the proposed method approaches to that of the known-channel case in terms of bit-error rates after the Viterbi decoder in overall SNRs.