• The Journal of the Acoustical Society of Korea
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• v.33 no.6
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• pp.400-406
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• 2014

This paper describes a robust watermark decoding model using a SVM(Support Vector Machine). First, the embedding process is performed inversely for a watermarked signal. And then the watermark is extracted using the proposed model. For SVM training of the proposed model, data are generated that are watermarks extracted from sounds containing watermarks by four different embedding schemes. BER(Bit Error Rate) values of the data are utilized to determine a threshold value employed to create training set. To evaluate the robustness, 14 attacks selected in StirMark, SMDI and STEP2000 benchmarking are applied. Consequently, the proposed model outperformed previous method in PSNR(Peak Signal to Noise Ratio) and BER. It is noticeable that the proposed method achieves BER 1% below in the case of PSNR greater than 10 dB.

• The Journal of the Korea institute of electronic communication sciences
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• v.15 no.6
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• pp.1037-1044
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• 2020

Since radar wind profiler (RWP) provides wind field data with high time and space resolution in all weather conditions, their verification of the accuracy and quality is essential. The simultaneous wind measurement from rawinsonde is commonly used to evaluate wind vectors from RWP. In this study, the simulation algorithm which produces the spectrum and raw time series (I/Q) data from the average values of moments is presented as a step-by-step verification method for the signal processing algorithm. The possibility of the simulation algorithm was also confirmed through comparison with the raw data of LAP-3000. The Doppler power spectrum was generated by assuming the density function of the skew-normal distribution and by using the moment values as the parameter. The simulated spectrum was generated through random numbers. In addition, the coherent averaged I/Q data was generated by random phase and inverse discrete Fourier transform, and raw I/Q data was generated through the Dirichlet distribution.

• 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.

• Journal of the Institute of Convergence Signal Processing
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• v.6 no.1
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• pp.39-44
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• 2005

Scheduling for digital system synthesis is assigning each operation in a control/data flow graph(CDFG) to a specific control step without violating precedence relation. It is one of the most important tasks due to its direct influence on the performance of the hardware synthesized. In this paper, we propose a resource-constrained scheduling algorithm. Our algorithm first analyzes the given CDFG to determine the number of functional units of each type, then assigns each operation to a control step while satisfying the constraints. It also tries to improve the solution iteratively by adjusting the number of functional units using the results collected from the previous scheduling. Experiments were performed to test the performance of the proposed algorithm, and results are presented

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.4
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• pp.303-309
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• 2013

A low power 6-bit flash ADC that uses an input voltage range detection algorithm is described. An input voltage level detector circuit has been designed to overcome the disadvantages of the flash ADC which consume most of the dynamic power dissipation due to comparators array. In this work, four digital input voltage range detectors are employed and each input voltage range detector generates the specific clock signal only if the input voltage falls between two adjacent reference voltages applied to the detector. The specific clock signal generated by the detector is applied to turn the corresponding latched comparators on and the rest of the comparators off. This ADC consumes 68.82mW with a single power supply of 1.2V and achieves 4.9 effective number of bits for input frequency up to 1MHz at 500 MS/s. Therefore it results in 4.75pJ/step of Figure of Merit (FoM). The chip is fabricated in 0.13-um CMOS process.

• Journal of Korea Society of Industrial Information Systems
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• v.21 no.3
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• pp.1-6
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• 2016

As the convergence speed of time domain adaptive algorithm on the LMS(Least Mean Square) becomes slow when eigen value distribution width is spread, So variable step size algorithm is used widely. But it needs a lot of calculation load. In this paper we consider new algorithm, which can reduce calculations and improve convergence speed, uses instantaneous absolute value of average noise signal adapting the exponential function. For the performance of proposed algorithm is tested and simulated to system generator. As the result we show the variable step size adaptive algorithm in proportion to instantaneous absolute value is more stable and efficient than others.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.15 no.6
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• pp.2128-2147
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• 2021

Cognitive radio systems are being implemented recently to tackle spectrum underutilization problems and aid efficient data traffic. Spectrum sensing is the crucial step in cognitive applications in which cognitive user detects the presence of primary user (PU) in a particular channel thereby switching to another channel for continuous transmission. In cognitive radio systems, the capacity to precisely identify the primary user's signal is essential to secondary user so as to use idle licensed spectrum. Based on the inherent capability, a new spectrum sensing technique is proposed in this paper to identify all types of primary user signals in a cognitive radio condition. Hence, a spectrum sensing algorithm using improved convolutional neural network and long short-term memory (CNN-LSTM) is presented. The principle used in our approach is simulated annealing that discovers reasonable number of neurons for each layer of a completely associated deep neural network to tackle the streamlining issue. The probability of detection is considered as the determining parameter to find the efficiency of the proposed algorithm. Experiments are carried under different signal to noise ratio to indicate better performance of the proposed algorithm. The PU signal will have an associated modulation format and hence identifying the presence of a modulation format itself establishes the presence of PU signal.

• Journal of Power System Engineering
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• v.6 no.3
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• pp.64-70
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• 2002

A fast decoupled algorithm for time domain simulation of power electronics circuits is presented. The circuits can be arbitrarily configured and can incorporate feedback amplifier circuits. This simulation algorithm is performed for the input series output parallel connected 2 switch forward converter. Steady state and large signal transient responses due to a step load change are simulated. The simulation results are verified through experiments.

• Proceedings of the IEEK Conference
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• 1999.06a
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• pp.187-190
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• 1999

In order to predict the effect of power amplifier non-linearity for digital modulated signal, this paper analyses the adjacent channel power ratio(ACPR) with the various AM-to-PM distortion levels. As the phase distortion increases from 0$^{\circ}$ to 12$^{\circ}$ at 1㏈compression point by 2.4$^{\circ}$ step, the input power level which satisfies the required ACPR decreases from 3.5㏈ to 6.5㏈ less than the 1㏈ compression input power.

• Journal of the Institute of Electronics Engineers of Korea SP
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• v.46 no.6
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• pp.95-101
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• 2009

Among the adaptive noise cancellers (ANC), the least mean square (LMS) algorithm has probably become the most popular algorithm because of its robustness, good tracking properties, and simplicity of implementation. However, it has non-uniform convergence and a trade-off between the rate of convergence and excess mean square error (EMSE). To overcome these shortcomings, a number of variable step size least mean square (VSSLMS) algorithms have been researched for years. These LMS algorithms use a complex variable step method approach for rapid convergence but need high computational complexity. A variable step approach can impair the simplicity and robustness of the LMS algorithm. The proposed instantaneous gain control (IGC) algorithm uses the instantaneous gain value of the original signal and the noise signal. As a result, the IGC algorithm can reduce computational complexity and maintain better performance.