• The Journal of Korean Institute of Communications and Information Sciences
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• v.22 no.7
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• pp.1427-1436
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• 1997

Turbo codes, decoded through an iterative decoding algorithm, habe recently been shown to yidel remarkable coding gains close to theoretical limits in the Gaussian channel environment. This thesis presents the performance of Turbo code through the computer simulation. The performance of modified Turbo code is compared to that of the conventional Turbo codes. The modified Turbo code reduces the time delay in decoding with minimal effect to the performance for voice transmission sytems. To achieve the same performance, random interleaver the size of which is no less than the square root of the original one should be used. Also, the modified Turbo code is applied to MC-CDMA system, and its performance is analyzed under the Rayleigh Fading channel environment. In Rayleigh fading channel environment, due to the amplitude distortion caused by fading, the interleaver of the size twice no less than that in the Gaussian channel enironment was required. In overall, the modified Turbo code maintained the performance of the conventional Turbo code while the time delay in transmission and decoding was reduced at the rate of multiples of two times the squared root of the interleaver size.

• Proceedings of the IEEK Conference
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• 1998.06a
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• pp.98-101
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• 1998

The intermodulation distortion(IMD) due to laser diode nonlinearity of an asynhcronous direct sequence code diviion multiple access(DS/CDMA) system in wireless-optical transmission system model is analzed. A third order polynomial is used to represent laser diode nonlinearity. In DS/CDMA system, only one harmonic of the third-order intermodulation term falls on the signal frequency band and influences the system performance characteristics. To cancel multi-user interference and nonlinear distortion in a DS/CDMA wireless-optical transmission system model, the simple transversal filter structure with N-taps of (N-1) tap delay of 1 chip time delay line is used. It is necessary to select an optimal modulation index that provides a maximum signal-to-noise ratio and the results are useful for CDMA system design.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.21 no.10
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• pp.1143-1153
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• 2010

In this paper, we propose an alternative method to increase the distortion cancellation bandwidth of an analog RF feedback power amplifier by using a negative group delay circuit(NGDC). A limited distortion cancellation bandwidth due to the group delay(GD) mismatch discouraged the use of feedback technique in spite of its powerful linearization performance. With the fabricated NGDC with positive phase slope over frequency, the feedback amplifier of the proposed topology experimentally achieved adjacent channel leakage ratio(ACLR) improvement of 15 dB over 50 MHz bandwidth at wideband code division multiple access(WCDMA) downlink band when tested with 2-carrier WCDMA signal. At an average output power of 28 dBm, ACLR of 25.1 dB is improved to obtain -53.2 dBc at 5 MHz offset.

• Journal of Satellite, Information and Communications
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• v.5 no.1
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• pp.54-57
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• 2010

The navigation system like GPS which is core technology is based on Code Division Multiple Access(CDMA) techniques. To receive satellite signal smoothly in CDMA, received signals have to synchronize with spread code. In this paper, we focus on the code tracking methods among synchronization techniques. The conventional delay lock loop(DLL) is unsuitable for multi-path channel. We will introduce how it overcomes distortion by multi-path. We will propose method that separates out multi-path signals and tracks the each path signals. And we will confirm performance of proposed method using Spirent simulator.

• Journal of Positioning, Navigation, and Timing
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• v.12 no.3
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• pp.237-244
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• 2023

A signal generation simulator is an economical and useful solution in Global Navigation Satellite System (GNSS) receiver design and testing. A software-defined radio approach is widely used both in receivers and simulators, and its flexible structure to adopt to new signals is ideally suited to the testing of a receiver and signal processing algorithm in the signal design phase of a new satellite-based navigation system before the deployment of satellites in space. The generation of highly accurate delayed sampled codes is essential for generating signals in the simulator, where its sampling rate should be chosen to satisfy constraints such as Nyquist criteria and integer and non-commensurate properties in order not to cause any distortion of original signals. A high sampling rate increases the accuracy of code delay, but decreases the computational efficiency as well, and vice versa. Therefore, the selected sampling rate should be as low as possible while maintaining a certain level of code delay accuracy. This paper presents the lower limits of the sampling rate for GNSS signal generation simulators. In the simulation, two distinct code generation methods depending on the sampling position are evaluated in terms of accuracy versus computational efficiency to show the lower limit of the sampling rate for several GNSS signals.

• Journal of Positioning, Navigation, and Timing
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• v.3 no.2
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• pp.37-44
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• 2014

The most effective method to overcome GPS jamming problem is to use an adaptive array antenna which has the capability of beamforming or nulling to a certain direction. In this paper, Space Time Adaptive Processing (STAP) beamforming algorithm of four elements array antenna will be designed and the anti-jamming performance will be analyzed. According to the analysis, the signal to noise ratio (SNR) and anti-jamming performance can be enhanced by beamforming algorithm. Also, the time tap effect of STAP algorithm will be analyzed theoretically and verified with array antenna modeling and simulation. Specially, the cautious selection of reference time tap in STAP can prevent the degradation of position accuracy performance.

• Journal of Advanced Marine Engineering and Technology
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• v.40 no.1
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• pp.45-49
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• 2016

The performance of underwater acoustic communication systems is sensitive to inter-symbol interference caused by delay spread developed from multipath signal propagation. The multipath nature of underwater channels causes signal distortion and error floor. In order to improve the performance, it is necessary to employ an iterative coding scheme. Of the various iterative coding schemes, turbo code and convolutional code based on the BCJR algorithm have recently dominated this application. In this study, the performance of iterative codes based on turbo equalizers with equivalent coding rates and similar code word lengths were analyzed. Underwater acoustic communication system experiments using these two coding techniques were conducted on Kyeong-chun Lake in Munkyeong City. The distance between the transmitter and receiver was 400 m, and the data transfer rate was 1 Kbps. The experimental results revealed that the performance of turbo codes is better for channeling than that of convolutional codes that use a BCJR decoding algorithm.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.11
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• pp.2295-2303
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• 2009

Underwater acoustic(UWA) communication has multipath error because of reflection by sea-level and sea-bottom. The multipath of UWA channel causes signal distortion and error floor. In this paper, we proposed the compensation method of multipath effect using the impulse response of the UWA channel and then analysis the performance of channel coding such as LDPC code, concatenate code. Also we analysed the time-delay errors and estimated amplitude errors of estimated channel information and its affection on the performance. As shown in simulation results, the performance of proposed compensation method is better than the performance of conventional method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.9B
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• pp.1742-1754
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• 1999

In this paper, we propose a rate control scheme, using histogram based rate-distortion (R-D) estimation, which produces a consistent picture quality between consecutive frames. The histogram based R-D estimation used in our rate control scheme offers a closed-form mathematical model that enable us to predict the bits and the distortion generated from an encoded frame at a given quantization parameter (QP) and vice versa. The most attractive feature of the R-D estimation is low complexity of computing the R-D data because its major operation is just to obtain a histogram or weighted histogram of DCT coefficients from an input picture. Furthermore, it is accurate enough to be applied to the practical video coding. Therefore, the proposed rate control scheme using this R-D estimation model is appropriate for the applications requiring low delay and low complexity, and controls the output bit-rate ad quality accurately. Our rate control scheme ensures that the video buffer do not underflow and overflow by satisfying the buffer constraint and, additionally, prevents quality difference between consecutive frames from exceeding certain level by adopting the distortion constraint. In addition, a consistent considering the maximum tolerance BER of the voice service. Also in Rician fading channel of K=6 and K=10, considering CLP=$10^{-3}$ as a criterion, it is observed that the performance improment of about 3.5 dB and 1.5 dB is obtained, respectively, in terms of $E_b$/$N_o$ by employing the concatenated FEC code with pilot symbols.