• The Journal of the Acoustical Society of Korea
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• v.41 no.4
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• pp.410-418
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• 2022

Channel reliability estimation for iterative codes such as turbo codes is very important factor in time varying underwater acoustic channel, an incorrect estimation of channel reliability induced performance degradation. Therefore, this paper presents an optimal channel reliability estimation method for turbo coded FSK signal with rate of 1/3. The estimated BER algorithm is a method that can estimate the reliability of received data by comparing received data and decoded data, and we determine optimal channel reliability by using the method. In order to analyze the performance, the experiment was conducted on a lake in Munkyeong city by moving in the range of 300 m to 500 m. At the result of applying presented method to failed decoding packets, we confirm all packets are decoded successfully.

• International Journal of Computer Science & Network Security
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• v.23 no.1
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• pp.96-102
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• 2023

In this paper, we proposed Channel Transfer Function estimation based on Delay and Doppler Profile for underwater acoustic OFDM communication system. It improved the estimation accuracy of the channel transfer function by linear time interpolation the change of Scattered Pilot (SP) insertion frequency in the time direction and the time by Delay and Doppler profile that analyzes the multipath situation of the channel investigated the performance of interpolation by simulation and report it. Previous works is inserted SP every 4 OFDM. It was effective under the environment without multipath, but it has observed that the effect of CTF compensation has been lowered in multipath channel condition. In addition to be better when inserted SP every 2 OFDM. But the amount of sending data will be decrease. Therefore, we conducted research to improve 4 OFDM with new interpolator. A computer simulation was performed as a comparison of SP inserted every 4 OFDM, SP inserted every 2 OFDM, and 4 OFDM with new interpolator. the performance of the proposed system is overwhelmingly improved, and the performance is slightly improved even 64 QAM.

• The Journal of the Acoustical Society of Korea
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• v.22 no.7
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• pp.573-578
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• 2003

Recently, speaker localizing estimation technique has been rising in teleconference systems. In this paper, it was described to be able to enhance the speech quality through microphone array, and received the only signal of speaker. Unfortunately, as it using estimated the signal in advance, it is not matched in a real acoustic environment so it has poor performance. In this paper is proposed for Adaptive Matched Filter Microphone Array that estimated acoustic room environment from the received the signal and study of the efficiency through simulations.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2007.10a
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• pp.165-169
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• 2007

Due to the complexity of underwater acoustic channel, signal estimation in underwater acoustic communication field is considerably affected from time-varying multipath fading channels. On this reason, the original signals should have many long training signals to estimate the channel and the purposed signals, and the bit rate of signals having information may have small rate. In order to avoid this loss of efficiency in underwater communication, this paper employed a blind channel identification method which don't use training signals. Simulations have predicted performance of the employed method in multipath environment and an aquatic plant experiment has verified the simulation results.

• International Journal of Naval Architecture and Ocean Engineering
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• v.7 no.2
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• pp.426-434
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• 2015

In this paper we proposed a dada-aided Doppler estimation method for underwater acoustic communication. The training sequence is non-dedicate, hence it can be designed for Doppler estimation as well as channel equalization. We assume the channel has been equalized and consider only flat-fading channel. First, based on the training symbols the theoretical received sequence is composed. Next the least square principle is applied to build the objective function, which minimizes the error between the composed and the actual received signal. Then an iterative approach is applied to solve the least square problem. The proposed approach involves an outer loop and inner loop, which resolve the channel gain and Doppler coefficient, respectively. The theoretical performance bound, i.e. the Cramer-Rao Lower Bound (CRLB) of estimation is also derived. Computer simulations results show that the proposed algorithm achieves the CRLB in medium to high SNR cases.

• The Journal of the Acoustical Society of Korea
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• v.35 no.5
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• pp.383-388
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• 2016

We develop two methods to select a constant in the RLS (Recursive Least Squares) with the convex regularization. The RLS with the convex regularization was proposed by Eksioglu and Tanc in order to estimate the sparse acoustic channel. However the algorithm uses the regularization constant which needs the information about the true channel response for the best performance. In this paper, we propose two methods to select the regularization constant which don't need the information about the true channel response. We show that the estimation performance using the proposed methods is comparable with the Eksioglu and Tanc's algorithm.

• The Journal of the Acoustical Society of Korea
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• v.27 no.3E
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• pp.69-76
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• 2008

Measurements of bubble size and distribution in the surface layer of the sea, wind speed, and variation of ocean environments were made continually over a four-day period in an experiment conducted in the South Sea of Korea during 17-20 September 2007. Theoretical background of bubble population model indicates that bubble population is a function of the depth, range and wind speed and bubble effects on sound speed shows that sound speed varies with frequency. Observational evidence exhibited that the middle size bubble population fit the model very well, however, smaller ones can not follow the model probably due to their short lifetime. Meanwhile, there is also a hysteresis effect of void fraction. Observational evidence also indicates that strong changes in sound speed are produced by the presence of swarms of micro bubbles especially from 7 kHz to 50 kHz, and calculation results are consistent with the measured data in the high frequency band, but inconsistent in the low frequency band. Based on the measurements of the sound speed and high frequency transmission configuration in the bubble layer, we present an estimation of underwater acoustic channel capacity in the bubble layer.

• The Journal of the Acoustical Society of Korea
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• v.38 no.3
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• pp.283-289
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• 2019

RLS (Recursive-least-squares) algorithm is known to have good convergence and excellent error level after convergence. However, there is a disadvantage that numerical instability is included in the algorithm due to inverse matrix calculation. In this paper, we propose an algorithm with no matrix inversion to avoid the instability aforementioned. The proposed algorithm still keeps the same convergence performance. In the proposed algorithm, we adopt an averaged gradient-based step size as a self-adjusted step size. In addition, a variable forgetting factor is introduced to provide superior performance for time-varying channel estimation. Through simulations, we compare performance with conventional RLS and show its equivalency. It also shows the merit of the variable forgetting factor in time-varying channels.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.1
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• pp.187-192
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• 2015

Underwater acoustic communication channels have very complex channel characteristics caused by time-varying sea surface, submarine topography, sound speed, and geometry between transmitter and receiver. Especially, the channel has time-variance and doppler effect due to wind and sea current. We have to recognize the channel state and apply it to communication technique for increasing transmission efficiency in the underwater acoustic channel. In this paper, we present the frame recursive modulation and demodulation method using ambiguity function and autocorrelation function to estimate the doppler frequency. Furthermore, we conducted the simulation and sea experiment to evaluate the performance of the proposed method. When the channel coding technique was not used, the bit error rate performance of the proposed method was improved about 32 % compared with conventional method.

• The Journal of the Acoustical Society of Korea
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• v.39 no.1
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• pp.32-37
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• 2020

This paper proposes a new l₁-norm-Recursive Least Squares (RLS) algorithm which is numerically more robust than the conventional l₁-norm-RLS. The l₁-norm-RLS was proposed by Eksioglu and Tanc in order to estimate the sparse acoustic channel. However the algorithm has numerical instability in the inverse matrix calculation. In this paper, we propose a new algorithm which is robust against the numerical instability. We show that the proposed method improves stability under several numerically erroneous situations.