• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.6
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• pp.1323-1329
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• 2014

In this paper, we analyzed efficient decoding scheme with FTN (Faster than Nyquist) method that is transmission method faster than Nyquist theory and increase the throughput. we proposed viterbi equalizer model to minimize ISI (Inter-Symbol Interference) when FTN signal is transmitted. the proposed model utilized interference as branch information. In this paper, to decode FTN singal, we used turbo equalization algorithms that iteratively exchange probabilistic information between soft Viterbi equalizer (BCJR method) and LDPC decoder. By changing the trellis diagram in order to maximize Euclidean distance, we confirmed that performance was improved compared to conventional methods as increasing throughput of FTN signal.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.8A
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• pp.725-732
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• 2002

Block-wise MAP (Maximum A posteriori) decoding algorithm for turbo-codes requires less memory than Log-MAP decoding algorithm. The ER (Bit Error Rate) performance of previous block-wise MAP decoding algorithm depend on the block length and training length. To maximize hardware utilization and perform successive decoding, the block length is set to be equal to the training length in previous MAP decoding algorithms. Simulation result on the BER performance shows that the EBR performance can be maintained with shorter blocks when training length is sufficient. This paper proposes an architecture for area efficient block-wise MAP decoder. The proposed architecture employs the decoding schema for reducing memory by using the training length, which in N times larger than block length. To efficiently handle the proposed schema, a pipelined architecture is proposed. Simulation results show that memory usage can be reduced by 30%~45% in the proposed architecture without degrading the BER performance.

• The Journal of the Acoustical Society of Korea
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• v.40 no.4
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• pp.370-381
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• 2021

This paper presented an efficient transceiver structure of multiband Frequency Shift Keying (FSK) signals with direct sequence spread spectrum for maintaining covertness and performance. In aspect to covertness, direct sequence spread spectrum method, which multiplying by Pseudo Noise (PN) codes whose rate is much higher than that of data sequence, is employed. In aspect to performance, in order to overcome performance degradation caused by multipath and Doppler spreading, we applied multiband, turbo equalization, and weighting algorithm are applied. Based on the simulation results, by applying 4 number of multiband and number of chips are 8 and 32, experiments were conducted in a lake with a distance of moving from 300 m to 500 m between the transceivers. we confirmed that the performance was improved as the number of bands and chips are increased. Furthermore, the performance of multiband was improved when the proposed weighting algorithm was applied.

• The Journal of the Acoustical Society of Korea
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• v.41 no.2
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• pp.218-226
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• 2022

The main characteristics of the underwater acoustic channel have varying multipath and fast fading. Especially, Frequency Shift Keying (FSK) signals with multiple frequency bands caused loss of information bits allocated on the specific frequencies due to selective fading phenomenon. Therefore, this paper proposes frequency gain control algorithm based on preamble error rates. The proposed algorithm estimates optimal gain value in the range of preamble error rate with less than 10 % for specific faded frequency. By employing turbo equalized FSK signals with rate of 1/3, the experiment was conducted on a lake in Munkyeong city with distance of 300 m to 500 m. The result confirms that packets are decoded successfully by applying proposed algorithm as increasing number of iterations.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.47 no.6
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• pp.66-74
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• 2010

Recently distributed video coding (DVC) is spotlighted for the environment which has restriction in computing resource at encoder. Wyner-Ziv (WZ) coding is a representative scheme of DVC. The WZ encoder independently encodes key frame and WZ frame respectively by conventional intra coding and channel code. WZ decoder generates side information from reconstructed two key frames (t-1, t+1) based on temporal correlation. The side information is regarded as a noisy version of original WZ frame. Virtual channel noise can be removed by channel decoding process. So the performance of WZ coding greatly depends on the performance of channel code. Among existing channel codes, Turbo code and LDPC code have the most powerful error correction capability. These channel codes use stochastically iterative decoding process. However the iterative decoding process is quite time-consuming, so complexity of WZ decoder is considerably increased. Analysis of the complexity of LPDCA with real video data shows that the portion of complexity of LDPCA decoding is higher than 60% in total WZ decoding complexity. Using the HDA (Hard Decision Aided) method proposed in channel code area, channel decoding complexity can be much reduced. But considerable RD performance loss is possible according to different thresholds and its proper value is different for each sequence. In this paper, we propose an adaptive HDA method which sets up a proper threshold according to sequence. The proposed method shows about 62% and 32% of time saving, respectively in LDPCA and WZ decoding process, while RD performance is not that decreased.

• 전자공학회논문지 IE
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• v.43 no.4
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• pp.115-121
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• 2006

In this paper, we proposed about data error check and correction on channel transmission in the communication system. LDPC codes are used for minimizing channel errors by modeling AWGN Channel as a VDSL system. Because LDPC Codes use low density parity bit, mathematical complexity is low and relating processing time becomes shorten. Also the performance of LDPC code is better than that of turbo code in long code word on iterative decoding algorithm. This algorithm is better than conventional algorithms to correct errors, the proposed algorithm assigns weights for errors concerning parity bits. The proposed weighted Bit-flipping algorithm is better than the conventional Bit-flipping algorithm and we are recognized improve gain rate of 1 dB.

• Journal of Navigation and Port Research
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• v.39 no.4
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• pp.293-297
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• 2015

Underwater acoustic communications has been limited use for military purposes in the past. However, the fields of underwater applications expend to detection, submarine and communication in recent. The excessive multipath encountered in underwater acoustic communication channel is creating inter symbol interference, which is limiting factor to achieve a high data rate and bit error rate performance. To improve the performance of a received signal in underwater communication, many researchers have been studied for channel coding scheme with excellent performance at low SNR. In this paper, we applied BCJR decoder based ( 2,1,7 ) convolution codes and to compensate for the distorted data induced by the multipath, we applying the turbo equalization method. Through the underwater experiment on the Gyeungcheun lake located in Mungyeng city, we confirmed that turbo equalization structure of BCJR has better performance than hard decision and soft decision of Viterbi decoding. We also confirmed that the error rate of decoder input is less than error rate of $10^{-1}$, all the data is decoded. We achieved sucess rate of 83% through the experiment.

• Proceedings of the IEEK Conference
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• 2005.11a
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• pp.423-426
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• 2005

In this paper, we propose a receiver that combines a channel detector with a channel decoder to retrieve information from ISI and AWGN in an iteratively manner. The receiver, evolving from a system of a PRML detector and a RS decoder, consists of a SOVA detector followed by a LDPC decoder and has them exchange information iteratively. Rather than handling extrinsic reliabilities explicitly as in Turbo equalization, we take hard-decision values from the LDPC decoder and mix them with the channel output in a certain ratio as input for SOVA. The scheme, simply modified to the one-way structure of a SOVA and a LDPC decoder, shows improved performance with iteration numbers as well as the combining ratio of the channel output and the feedback output. We additionally analyze the receiver with a simple theoretical model and present some valuable properties.

• Transactions of the Korean Society for Noise and Vibration Engineering
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• v.15 no.6 s.99
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• pp.687-696
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• 2005

Acoustic power spectrum of the upstream and downstream sound field due to an isotropic frozen turbulent gust impinging on a cascade of flat plate airfoils are computed by using a analytic formulation derived from Smith's method, and Whitehead's LINSUB codes. A parametric study of the effects on sound power of the number of blades and turbulence length scale is performed with an emphasis on analyzing the characteristics of sound power spectrum. Through the comparison of the computed results of sound power, it is found that acoustic power spectrum from the 2-D cascade subject to a ingested turbulence can be categorized into two distinct regions. one is lower frequency region where some spectral components of turbulence do not contribute to the cut-on acoustic modes and therefore the effect of the cascade geometry is more dominant ; the other is higher frequency region where all of spectral components of turbulence make contributions to cut-on acoustic modes and thus acoustic power is approximately proportional to the blade number.

• IEIE Transactions on Smart Processing and Computing
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• v.3 no.3
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• pp.142-152
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• 2014

Vehicular Ad-hoc Networks (VANETs) are comprised of wireless mobile nodes characterized by a randomly changing topology, high mobility, availability of geographic position, and fewer power constraints. Orthogonal Frequency Division Multiplexing (OFDM) is a promising candidate for the physical layer of VANET because of the inherent characteristics of the spectral efficiency and robustness to channel impairments. The susceptibility of OFDM to Inter-Carrier Interference (ICI) is a challenging issue. The high mobility of nodes in VANET causes higher Doppler shifts, which results in ICI in the OFDM system. In this paper, a frequency domain com-btype channel estimation was used to cancel out ICI. The channel frequency response at the pilot tones was estimated using a Least Square (LS) estimator. An efficient interpolation technique is required to estimate the channel at the data tones with low interpolation error. This paper proposes a robust interpolation technique to estimate the channel frequency response at the data subcarriers. The channel induced noise tended to degrade the Bit Error Rate (BER) performance of the system. Parallel concatenated Convolutional codes were used for error correction. At the decoding end, different decoding algorithms were considered for the component decoders of the iterative Turbo decoder. A performance and complexity comparison among the various decoding algorithms was also carried out.