• International Journal of Naval Architecture and Ocean Engineering
• /
• v.9 no.3
• /
• pp.256-265
• /
• 2017

In this paper, an ascheme is proposed for multiuser underwater acoustic communication by using the multi-chirp rate signals. It differs from the well known TDMA (Time Division Multiple Access), FDMA (Frequency Division Multiple Access) or CDMA (Code Division Multiple Access), by assigning each users with different chirp-rate carriers instead of the time, frequency or PN code. Multi-chirp rate signals can be separated from each other by FrFT (Fractional Fourier Transform), which can be regarded as the chirp-based decomposing, and superior to the match filter in the underwater acoustic channel. VTRM (Virtual Time Reverse Mirror) is applied into the system to alleviate the ISI caused by the multipatch and make the equalization more simple. Results of computer simulations and pool experiments prove that the proposed multiuser underwater acoustic communication based on the multi-chirp rate exhibit well performance. Outfield experments carrie out in Xiamen Port show that using about 10 kHz bandwidth, four users could communicate at the same time with 425 bps with low BER and can match the UAC application.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.26 no.12A
• /
• pp.1978-1982
• /
• 2001

In a Multi-carrier modulation system, CP (Cyclic prefix) is inserted in the transmit tame in order to eliminate the ISI (Intersymbol Interference) and ICI (Interchannel Interference) caused by delay spread of a received signal, which in rum degrades the throughput of the system. TEQ (Time-domain equalizer) improves the system throughput by shortening the CIR (Channel Impulse Response) time and maintaining the CP length to the minimum regardless of the channel condition. In this paper, a new MMSE (Minimum Mean Square Error) TEQ algorithm is proposed and its performance is analyzed in order to speed up computing the optimum tap coefficients of the equalizer by employing Newton method.

• The Proceeding of the Korean Institute of Electromagnetic Engineering and Science
• /
• v.5 no.3
• /
• pp.22-35
• /
• 1994

To improve the bandwidth efficiency in the environment of digital mobile communications, a correlative coded FM system is designed. The signal of this system has continuous phase and high power efficiency due to the constant envelope. But this signal also has a little loss of the SNR and some degradation of the BER. In this paper, a modified MLSE method which uses correlative coded signal is adopted to improve the performance of the receiver. The MLSE method improved the BER performance in the used channel. Without the decision feedback, the receiver performance was improved by 2dB and with it, by 4dB Particularly, the MLSE method and the decision feedback showed better performance in bad channels than in a stable telephone channel.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
• /
• v.11 no.2
• /
• pp.158-165
• /
• 2000

In this paper, the transmission channels of L-band and Ka-band multi-media satellite communication systems for the land mobile satellite(LMS) communication service are modeled. Trellis coded(TC) 8PSK is proposed as a power and bandwidth efficient digital transmission scheme for the LMS system, and its error probability performance is analyzed. Block interleaving and deinterleaving are applied to the transmitter and receiver of LMS system respectively in order to compensate for the BER performance degradation of TC-8PSK caused by multipath fading. Viterbi equalizer is also employed in the receiver for channel equalization, and the corresponding BER performance improvement is analyzed.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.7A
• /
• pp.659-667
• /
• 2002

In this dissertation, in the Turbo Code used for error correction coding of the recent digital communication systems, we propose a new S-R interleaver that has the better performance than the existing block interleaver, and the Turbo Decoder that has the parallel concatenated New structure using the MAP algorithm. For real-time voice and video services over the third generation mobile communications, the performance of two proposed methods is analyzed by the reduced decoding delay using the variable decoding method by computer simulation over multipath channels of DS-CDMA system. Also, a Modified NCMA based on conventional NCMA is proposed to improve the channel efficiency in the mobile communication system, and is investigated over the multi-user environment of DS-CDMA system through computer simulation.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.38A no.4
• /
• pp.318-324
• /
• 2013

For blind channel equalization, the reliability of signal estimate determines the convergence speed and steady-state performance of the equalizer. Therefore the nonlinear estimator and reference signal being used in signal estimate should be chosen appropriately. In this paper, to increase the reliability of the signal estimate, two errors were obtained by applying coarse signal points and dense signal points respectively to signal estimate, the relative reliabilities of two errors were calculated, then the equalizer was adapted deferentially utilizing the reliabilities. At this point, by applying two errors alternately, two modes of operation were smoothly combined. Through computer simulations the proposed method was confirmed to achieve fast transient state convergence and low steady-state error compared to traditional methods.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.25 no.3A
• /
• pp.375-382
• /
• 2000

Equalization of satellite communication using Complex-Bilinear Recurrent Neural Network(C-BLRNN) is proposed in this pater. Since the BLRNN is based on the bilinear polynomial and it has been more effectively used in modeling highly nonlinear systems with time-series characteristics than multi-layer perception type neural networks(MLPNN) , it can be applied to satellite equalizer. the proposed C-BLRNN based equalizer for M-PSK with a channel model is compared with Volterra filter Equalizer, DFE, and conventional Complex MLPNN Equlizer. The results show that the proposed C-BLRNN based equalizer gives very favorable results in both of MSE and BER criteria over other equalizers.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.26 no.7B
• /
• pp.937-943
• /
• 2001

fast fading 채널 특성을 갖는 DS-CDMA 다중 사용자 환경에서 Normalized CMA(Constant Modulus Algorithm)와 Newton 방식을 이용한 CMA를 이용하여 빠른 수렴속도와 작은 평균 자승 오차(Mean Square Error)를 동시에 개선할 수 있는 등화 방법을 제안하였다. Normalized CMA는 Newton 방식을 이용한 CMA에 비해 작은 평균 자승오차를 갖지만 수렴속도가 느리다는 단점이 있다. 반면 Newton 방식을 이용한 CMA는 Normalized CMA에 비해 수렴속도는 빠르지만 큰 평균 자승 오차를 갖는다는 단점이 있다. 따라서 빠른 수렴 속도와 작은 평균 자승 오차를 동시에 얻기 위한 구조를 제안하였으며, 이 구조는 각각의 알고리즘을 사용하는 방법과는 달리 두 개의 알고리즘을 동시에 이용한다. 모의 실험 결과, 제안한 기법이 Normalized CMA보다 약 320번, Newton 방식을 이용한 CMA보다는 170번 정도 빠른 수렴 속도를 나타냈으며, 동시에 수렴시의 평균 자승 오차는 Newton 방식을 이용한 CMA보다 약 0.6dB, Normalized CMA보다 약 0.4dB 정도 낮은 수치를 나타내는 것을 확인할 수 있었다.

• Proceedings of the Acoustical Society of Korea Conference
• /
• spring
• /
• pp.215-218
• /
• 2000

입체음향 생성을 위한 기존의 방법은 크게 바이노럴 녹음기법과 머리전달함수(HRTF)를 이용한 바이노럴 합성 기법으로 나눌 수 있다. 현재 바이노럴 기법은 기존 스테레오 시스템에 비해 공간감, 몰입감 측면에서는 탁월한 효과가 있지만, 음질의 저하와 정면 음상 정위가 잘되지 않는다는 치명적인 단점 때문에 프로페셔널 오디오 분야에서는 거의 사용되지 않고 몇몇 PC 게임용으로만 사용되고 있다. 본 논문은 정확한 정면 음상 정위를 위해 '3채널 더미헤드를 이용한 바이노럴 녹음기법' 을 제안하고, 기존 스테레오 녹음기법과의 호환성 유지를 위해 녹음 현장에서 직접 사용될 수 있는 3채널 더미헤드를 사용한 'Weighted Diffuse-field equalization 기법'에 대해 제안하며, 3 채널 더미헤드를 이용하여 기존 HRTF 데이터를 대체할 수 있는 정면 음상 정위에 강인한 '3 채널 더미헤드 HRTF 측정 기법'에 대해 제안한다.

• The Journal of the Acoustical Society of Korea
• /
• v.28 no.3E
• /
• pp.93-98
• /
• 2009

In this paper, we propose a new VSS-NLMS algorithm through a simple modification of the conventional NLMS algorithm, which leads to a low complexity algorithm with enhanced performance. The step size of the proposed algorithm becomes smaller as the error signal is getting orthogonal to the input vector. We also show that the proposed algorithm is an approximated normalized version of the KZ-algorithm and requires less computation than the KZ-algorithm. We carried out a performance comparison of the proposed algorithm with the conventional NLMS and other VSS algorithms using an adaptive channel equalization model. It is shown that the proposed algorithm presents good convergence characteristics under both stationary and non-stationary environments despites its low complexity.