• The Journal of Korean Institute of Communications and Information Sciences
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• v.23 no.7
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• pp.1707-1717
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• 1998

Polyphase filtering techniques is used to suppress the narrowband jammer signal such as USDC TDMA overlaying the band occupied by DS-CDMA system. In the proposed jammer suppression, the received signal is separated into 64 subchannels in two stages by polyphase filtering and the location of the narrowband jammer signal is determined by measuring each subchannel power and the contaminated subchannels are simply blocked. The $E_{b}/N_{0}$ 0/ improvement of the CDMA system from jammer suppession was outstanding. The $E_{b}/N_{0}$ degradation in comparison with a performance of no jammer is around 0.8dB in the worst case. The results are also compared with those of linear prediction jammer suppression. The implementation of the ployphase jammer suppression requires great amount of data processing and computation compared to linear predication filter. Thus it is more appropriate to implement with a ASIC rather than WITH several DSPs for user terminals of forward link.

• The Journal of the Acoustical Society of Korea
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• v.16 no.2
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• pp.41-48
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• 1997

This paper proposes the methods for improving the recognition rate of theARS, especially equipped with the speech recognition capability. Telephone speech, which is the input to the ARS, is usually affected by the announcements from the system, channel noise, and channel distortion, thus directly applying the recognition algorithm developed for clean speech to the noisy telephone speech will bring the significant performance degradation. To cope with this problem, this paper proposes three methods: 1)the accurate detection of the inputting instant of the speech in order to immediately turn off the announcements from the system at that instant, 2)the effective end-point detection of the noisy telephone speech on the basis of Teager energy, and 3)the SDCN-based compensation of the channel distortion. Experiments on speaker-independent, noisy telephone speech reveal that the combination of the above three proposed methods provides great improvements on the recognition rate over the conventional method, showing about 77% in contrast to only 23%.

• The Journal of the Acoustical Society of Korea
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• v.34 no.4
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• pp.295-302
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• 2015

Acoustic channels are characterized by long multipath spreads that cause inter-symbol interference. The way in which this fact influences the design of the receiver structure is considered. To satisfy performance and throughput, we presented consecutive iterative BCJR (Bahl, Cocke, Jelinek, Raviv) equalization to improve the performance and throughput. To achieve low error performance, we resort to powerful BCJR equalization algorithms that iteratively update probabilistic information between inner decoder and outer decoder. Also, to achieve high throughput, we divide long packet into consecutive small packets, and the estimate channel information of previous packets are compensated to next packets. Based on experimental channel response, we confirmed that the performance is improved for long length packet size.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.3
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• pp.315-320
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• 2010

In this paper, an internal TDMB(Terrestrial Digital Multimedia Broadcasting) antenna for mobile phone is proposed. The overall dimension of designed antenna with substrate is 30 mm$\times$5 mm$\times$0.6 mm. The proposed antenna consists of a meander type radiator which is connected front- and back-plane of Kapton substrate by via hole and parasitic element for tuning the resonant frequency. And to compensate the electric length of desired frequency, passive inductor is used for matching element. Measured gain of the implemented antenna -17.6 dBi at 174 MHz, -13.01 dBi at 195 MHz, and -14.9 dBi at 216 MHz.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.1
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• pp.164-171
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• 2008

Adaptive equalizers are inevitable schemes in digital communication systems for compensating the transmission channel distortion. Additionally, to obtain the required BER(Bit Error Rate), the adaptive algorithms appropriate to the mobile communication channels are required. In this paper, we propose the NPDFE (Noise-Predictive Decision Feedback Equalizer) for communication systems performance improvement in mobile communication channels. The performance of the proposed NPDFE with QPSK (Quadrature Phase Shift Keying) is simulated under AWGN (Additive White Gaussian Noise), Ricean fading, ETSI (European Telecommunications Standards Institute) fading, and Rayleigh fading channels. The equalizers used in simulations are a LE (Linear Equalizer), a DFE (Decision Feedback Equalizer), and a NPDFE. Moreover, the equalizer performance criterion of the QPSK is the BER.

• The Journal of the Acoustical Society of Korea
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• v.37 no.6
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• pp.451-456
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• 2018

UAComm (Underwater Acoustic Communication) is an active research area, and many experiment has been performed to develop UAComm system. In this paper, we investigate the possibility of modifying and applying VirTEX (Virtual Time series EXperiment) to medium long range MIMO (Multiple-Input Multiple-Output) UAComm of about 20 km range for the analysis and performance prediction of UAComm system. Since VirTEX is a time-domain simulator, the generated time series can be used in HILS (Hardware In the Loop Simulation) to develop UAComm system. The developed package is verified through comparing with the sea-going FAF05 (Focused Acoustic Field 2005) experimental data. The developed simulator can be used to predict the performance of UAComm system, and even replace the expensive sea-going experiment.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4C
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• pp.413-423
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• 2003

The CMA algorithmis most widely investigated blind algorithm and the most widely used one in practice. But, since nonlinear CM cost function have not closed form solution about the optimum weight. There have been difficultiesto analyze the CMA equalizer's theoretical performance. Recently, Zeng presents the notable theoretical resultabout the MSE of CM-minimizing estimators for the FIR linear channel in the presence of AWGN. Through this method, It wouldbe possible to campare the theoretical performance between CMA and Wiener equalizer in terms of MSE. In this paper, based on Zeng's method, we first calculate the theoretical MSE bound of CMA equalizer in partial response channel which is widely used in HDD, digital VCR such as high-density digital recording.playback systems. We confirmedthis result withthe computer simulation. Except this, we also performedthe theoretical and simulation analysis about the modified CMA equalizer, which was proposed to improve the performance of CMA equalizer in partial response channel. Finally, we compare and evaluate the performance analysis results between CMA and Modified CMA equalizer.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.2
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• pp.443-449
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• 2015

In this paper, we proposed Channel allocation scheme according to the user's location with IR. In VLC System, LED can generate various colors of light by controlling the mixing ratio of each individual RGB color element. Thus, each RGB channel will have a different signal power, and each channel will have different performance. This proposed system using Visible light(RGB) as way to transmit signals, it depends on the mixture RGB, which decided the color of light, moreover, each things determined their performance. However, if the signal were fixed allocated RGB to transmit such as the original system, the importance of the each signals a different occur the limit on the quality of signals. To solve this problem in this paper, according to the RGB mixture ratios analyze the performance for the LED, which analyzed based on allocating the signal by transmitting to improve the quality was about how researched. In addition, our proposed system is able to improve the performance of BER and satisfied the Qos to desire users.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.503-506
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• 2001

In this paper, we describe the implementation of a microphone input level control algorithm and the speech improvement with this level controller in personal computer environment. The volume of speech obtained through a microphone affects the speech recognition rate directly. Therefore, proper input volume level control is desired fur better recognition. We considered some conditions for the successful volume controller implementation firstly, then checked its usefulness on our speech recognition system with common office environment speech database. Cepstral mean subtraction is also utilized far the channel-effect compensation of the database. Our implemented controller achieved approximately 50% reduction, i.e., improvement in speech recognition error rate.

• Proceedings of the KSPS conference
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• 2002.11a
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• pp.201-204
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• 2002

As a preliminary study for improving recognition performance of the connected digit telephone speech, we investigate feature parameters as well as channel compensation methods of telephone speech. The CMN and RTCN are examined for telephone channel compensation, and the MFCC, DWFBA, SSC and their delta-features are examined as feature parameters. Recognition experiments with database we collected show that in feature level DWFBA is better than MFCC and for channel compensation RTCN is better than CMN. The DWFBA+Delta_ Mel-SSC feature shows the highest recognition rate.