• Journal of the Institute of Convergence Signal Processing
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• v.5 no.3
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• pp.216-222
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• 2004

Frequency selective fading is a limiting factor for transmission rate and performance in high-speed multimedia communications. In this paper, we propose a turbo coded adaptive quadrature amplitude modulation (QAM) system for efficient high-speed transmission. By making use of a user-friendly simulation platform of SPW, the proposed turbo coded adaptive QAM system(TuAQAM) is developed and its performance is evaluated in terms of throughput and BER performance. Two channel models having delay spreads of 700ns and 1400ns are created for the simulations. It is shown that the proposed TuAQAM system provides a performance improvement of approximately 3dB and improved throughput for the channel model whose delay spread is 700ns. Similarly, a performance improvement is also achieved for the channel model whose delay spread is 1400ns.

• Journal of Electrical Engineering and Technology
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• v.10 no.4
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• pp.1655-1666
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• 2015

This paper proposes a new approach of Field Programmable Gate Array (FPGA) controlled digital implementation of shunt active power filter (SAPF) under steady state and dynamic operations. Typical implementations of SAPF uses microprocessor and digital signal processor (DSP) but it limited for complex algorithm structure, absence of feedback loop delays and their cost can be exceed the benefit they bring. In this paper, the hardware resources of an FPGA are configured and implemented in order to overcome conventional microcontroller or digital signal processor implementations. This proposed FPGA digital implementation scheme has very less execution time and boosts the overall performance of the system. The FPGA controller integrates the entire control algorithm of an SAPF, including synchronous reference frame transformation, phase locked loop, low pass filter and inverter current controller etc. All these required algorithms are implemented with a single all-on chip FPGA module which provides freedom to reconfigure for any other applications. The entire algorithm is coded, processed and simulated using Xilinx 12.1 ISE suite to estimate the advantages of the proposed system. The coded algorithm is also defused on a single all-on-chip Xilinx Spartan 3A DSP-XC3SD1800 laboratory prototype and experimental results thus obtained match with simulated counterparts under the dynamic state and steady state operating conditions.

• The Journal of the Acoustical Society of Korea
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• v.32 no.5
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• pp.377-384
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• 2013

In this paper, performance of COFDM (Coded Orthogonal Frequency Division Multiplexing) which is OFDM with a forward error correction code, is studied in frequency selective fading underwater acoustic communication channel. The OFDM is a multiplexing technique resistant to frequency selective multipath channel. In OFDM, a broadband information signal is transformed into several narrow band signals and transmits narrow band signals whose bandwidths are less than the channel coherence bandwidth. However, its performance is degraded in a specific narrow band signal due to its deep fading by multipath. To mitigate this degradation, COFDM which is OFDM with convolution code as a forward error correction code, is evaluated. The performance of COFDM is found to be better than that of OFDM in multipath channel.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2C
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• pp.117-122
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• 2006

In this paper, we present a simple and general soft bit decision expression for a Gray coded PAM signal over additive white Gaussian noise(AWGN) channel with the log likelihood ratio(LLR). In order to reduce the complexity of the LLR calculation, we make the bitwise LLR expression simple by replacing the mathematical max functions of the conventional Max-Log-MAP expression with simple arithmetic functions associated with some deterministic parameters, such as a received value and distance between symbols on a signal space. Taking the implementation issues, like the area of silicon, the power consumption, the timing latency, and so on, into consideration, we submit that the proposed method is a promising alternative way to conventional methods for reconfigurable systems.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.38 no.6
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• pp.93-105
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• 2001

In the lossless step of Lattice Vector Quantization(LVQ), the lattice codewords produced at quantization step are enumerated into radius sequence and index sequence. The radius sequence is run-length coded and then entropy coded, and the index sequence is represented by fixed length binary bits. As bit rate increases, the index bit linearly increases and deteriorates the coding performances. To reduce the index bits across the wide range of bit rates, we developed a novel lattice enumeration algorithm adopting the set partitioning method. The proposed enumeration method shifts down large index values to smaller ones and so reduces the index bits. When the proposed lossless coding scheme is applied to a wavelet based image coding, the proposed scheme achieves more than 10% at bit rates higher than 0.3 bits/pixel over the conventional lossless coding method, and yields more improvement as bit rate becomes higher.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2004.05b
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• pp.190-195
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• 2004

In this paper, Error rate performance of Convolution coded DS-CDMA 16 QAM signal is analyzed using selective combining diversity reception techniques and maximal ratio combining diversity reception techniques under the environments of Rician fading. With the results of analysis, maximal ratio combining diversity reception techniques provides the performance improvement of about 3-8[㏈] over selective combining diversity reception techniques for the good error performance lot data communication. And it is found that a synergistic performance improvement is show to both diversity reception and Convolution coding, techniques overcoming mobile wireless data communication channel environment.

• ETRI Journal
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• v.36 no.2
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• pp.242-252
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• 2014

Since July of 2012, the 3D video extension of H.264/AVC has been under development to support the multi-view video plus depth format. In 3D video applications such as multi-view and free-view point applications, synthesized views are generated using coded texture video and coded depth video. Such synthesized views can be distorted by quantization noise and inaccuracy of 3D wrapping positions, thus it is important to improve their quality where possible. To achieve this, the relationship among the depth video, texture video, and synthesized view is investigated herein. Based on this investigation, an edge noise suppression filtering process to preserve the edges of the depth video and a method based on a total variation approach to maximum a posteriori probability estimates for reducing the quantization noise of the coded texture video. The experiment results show that the proposed methods improve the peak signal-to-noise ratio and visual quality of a synthesized view compared to a synthesized view without post processing methods.

• Journal of Navigation and Port Research
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• v.29 no.3 s.99
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• pp.245-250
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• 2005

In this paper, considering PAR of transmitter which is pointed out OFDM system's problem, we designed Coded OFDM (COFDM) system and estimated BER and SNR using PAR reduction method In order to evaluate performance, we compared M-ary PSK (M-ary Phase Shift Keying) with M-ary QAM (M-ary Quadrature Amplitude Modulation). In result, performance of 16-PSK and 16-QAM came to good Moreover, 16-QAM showed better performance of about 2 dB in 10-3 error probability and performance of about 5 dB in Peak power clipping than that of 16-PSK.

• Journal of Communications and Networks
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• v.7 no.3
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• pp.243-247
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• 2005

We propose an efficient block least-mean-square (BLMS) adaptive algorithm, in conjunction with error control coding, for direct-sequence code division multiple access (DS-CDMA) systems. The proposed adaptive receiver incorporates decision feedback detection and channel encoding in order to improve the performance of the standard LMS algorithm in convolutionally coded systems. The BLMS algorithm involves two modes of operation: (i) The training mode where an uncoded training sequence is used for initial filter tap-weights adaptation, and (ii) the decision-directed where the filter weights are adapted, using the BLMS algorithm, after decoding/encoding operation. It is shown that the proposed adaptive receiver structure is able to compensate for the signal-to­noise ratio (SNR) loss incurred due to the switching from uncoded training mode to coded decision-directed mode. Our results show that by using the proposed adaptive receiver (with decision feed­back block adaptation) one can achieve a much better performance than both the coded LMS with no decision feedback employed. The convergence behavior of the proposed BLMS receiver is simulated and compared to the standard LMS with and without channel coding. We also examine the steady-state bit-error rate (BER) performance of the proposed adaptive BLMS and standard LMS, both with convolutional coding, where we show that the former is more superior than the latter especially at large SNRs ($SNR\;\geq\;9\;dB$).

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.6
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• pp.140-145
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• 2017

A channel is very sever in an underwater acoustic communication. Therefore, a modulation method with high spectrum efficiency is needed in an underwater acoustic communication. PSK(phase shift keying) for transmitting 1 bit or 2 bits is robust to noise, but sensitive to noise about more than 3 bits. CPFSK error performance for transmitting 1 bit or 2 bits is similar to that of PSK and CPFSK decreases high frequency components in modulation signal. In the paper, I analyze the performance of trellis coded 4-CPFSK modulation.