• Journal of the Institute of Electronics Engineers of Korea SP
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• v.44 no.4 s.316
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• pp.109-116
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• 2007

Most harmonic synthesis methods using phase information employ a quadratic or cubic phase interpolation. The methods are computationally expensive to implement because every component sinewave must be synthesized on a per sample basis. In this paper, we propose a fast harmonic synthesis method for sinusoidal speech/audio coding based on the quadratic and cubic phase function to overcome the complexity problem. To derive the fast harmonic synthesis method, we define the over-sampling function and phase modulation function by constraining the parameter of phase function to be independent for harmonic index and derive the fast synthesis method using IFFT. Experimental results show that the proposed method significantly reduce the complexity of conventional cosine synthesis method while maintaining the performance.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.8
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• pp.10-15
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• 2009

In this paper, we propose a frequency-domain tone interference cancellation algorithm for direct-sequence spread spectrum systems. In the previously proposed frequency-domain interference cancellation algorithms, the interference signal is estimated and compensated in the frequency-domain and then, with IFFT (inverse Fourier transform), the compensated frequency-domain signal is transformed to the time domain signal for the remaining receiver operations. Unlike the previous works, the proposed algorithm does not requires IFFT and thus, the proposed algorithm not only shows the virtually identical performance but also requires lower hardware complexity compared to the previous works.

• Journal of Communications and Networks
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• v.1 no.2
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• pp.86-88
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• 1999

The Discrete Fourier Transform (DFT) and the Inverse Discrete Fourier Transform (IDFT) are commonly used in signal processing applications, in particular in digital communication sys-tems using the multi-carrier modulation principle. In such systems an IDFT is computed at the transmitter end, and a DFT at the re-ceiver end. This paper examines a technique of computations, for which only negligible differences appear between the DFT and the IDFT calculations while the number of arithmetic operations re-quired is substantially reduced. This offers significant advantages for the design of an IDFT/DFT processor for Discrete Multitone(DMT) systems.

• The Journal of the Acoustical Society of Korea
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• v.27 no.3
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• pp.140-148
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• 2008

In this study, a numerical method for a time-domain acoustic wave backscattering analysis is established based on a physical optics and a Fourier transform. The frequency responses of underwater targets are calculated based on physical optics derived from the Kirchhoff-Helmholtz integral equation by applying Kirchhoff approximation and the time-domain signals are simulated taking inverse fast Fourier transform to the obtained frequency responses. Particularly, the adaptive triangular beam method is introduced to calculate the areas impinged directly by acoustic incident wave and the virtual surface concept is adopted to consider the multiple reflection effect. The numerical analysis result for an acoustic plane wave field incident normally upon a square flat plate is coincident with the result by the analytic time-domain physical optics derived theoretically from a conventional physical optics. The numerical simulation result for a hemi-spherical end-capped cylinder model is compared with the measurement result, so that it is recognized that the presented method is valid when the specular reflection effect is predominant, but, for small targets, gives errors due to higher order scattering components. The numerical analysis of an idealized submarine shows that the established method is effectively applicable to large and complex-shaped underwater targets.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.2
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• pp.113-122
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• 2016

In this paper, we suggest a approach which comprises fast Fourier transform inversion by wavelet noise attenuation. It represents an inverse filtering by adopting a factor into the Wiener filtering, and the optimal factor is chosen to minimize the overall mean squared error. in order to apply the Wiener filter, we have to compute the power spectrum of original image from the corrupted figure. Since the Wiener filtering contains the inverse filtering process, it expands the noise when the blurring filter is not invertible. To remove the large noises, the best is to remove the noise using wavelet threshold. Wavelet noise attenuation steps are consisted of inverse filtering and noise reduction by Wavelet functions. experimental results have not outperformed the other methods over the overall restoration performance.

• Geophysics and Geophysical Exploration
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• v.16 no.3
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• pp.139-144
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• 2013

A computer program has been developed to estimate time-domain electromagnetic (EM) responses for a onedimensional model with multiple source and receiver dipoles that are finite in length. The time-domain solution can be obtained by applying an inverse fast Fourier transform (FFT) to frequency-domain fields for efficiency. Frequency-domain responses are first obtained for 10 logarithmically equidistant frequencies per decade, and then cubic spline interpolated to get the FFT input. In the case of phases, the phase curve must be made to be continuous prior to the spline interpolation. The spline interpolated data are convolved with a source current waveform prior to FFT. In this paper, only a step-off waveform is considered. This time-domain code is verified with an analytic solution and EM responses for a marine hydrocarbon reservoir model. Through these comparisons, we can confirm that the accuracy of the developed program is fairly high.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.6C
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• pp.565-571
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• 2007

In order to improve PAPR (Peak-to-Average Power Ratio) reduction performance of the conventional SLM (Selective Mapping) for OFDM (Orthogonal Frequency Division Multiplexing) signals, we propose an effective SLM-PRSC (PAPR Reduction Sub-Carrier) hybrid scheme. In the proposed scheme, after performing the SLM for the frequency domain OFDM symbol excluding pre-determined PRSC positions, the SLM-PRSC hybrid sequence with the lowest PAPR generated by adding the time domain PRSC sequence to the results of the SLM, is selected as the transmitted OFDM signal. Since the identical PRSC sequences generated a priori are repeatedly used for every OFDM symbol, excessive IFFT (inverse Fast Fourier Transform) calculation is avoided. Simulation results show that the proposed scheme significantly improves the PAPR reduction performance of the conventional SLM, while avoiding excessive increase of IFFT calculation and the overhead for the SLM.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.11 s.102
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• pp.1123-1130
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• 2005

In this paper, we propose an improved side information processing scheme which is important in the sub-block phase weighting(SPW) method for the peak-to-average power ratio(PAPR) reduction. SPW method is to divide the input OFDM subchannels into several subblocks and to multiply phase weighting with each subblocks, properly for the reduction of the peak power. SPW method is similar to the conventional PTS method when the number of sub-carriers, signal modulation format and the number of subblocks are the same. However, unlike the conventional PTS(Partial Transmit Sequence) and SLM(Selected Mapping) method using many stages of IFFT(Inverse Fast Fourier Transform), SPW method only needs one IFFT. Although PAPR can be reduced by SPW method, complex computation burden still remains. In this paper the flipping algorithm and the full iteration algorithm are used f3r the phase control method. Through the computer simulation, we analyze and discuss the properties and the performance of the suggested method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.12C
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• pp.1142-1148
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• 2007

In this paper, we propose ASLM (Adaptive Selective Mapping; ASLM) scheme based on peak observation for PAPR (Peak-to-Average Power Ratio) reduction of OFDM (Orthogonal Frequency Division Multiplexing) signals. The proposed scheme is composed of three steps: peak scaling, sequence selection, and SLM procedures. In the first step, the peak signal samples in the IFFT (Inverse Fast Fourier Transform) outputs of the original input sequence are scaled down. In the second step, the sub-carrier positions where the power difference between the original input sequence and the FFT output of the scaled signal is large, are identified. Then, the phase sequences having the maximum number of phase-reversed sequence words only for these positions are selected. Finally, the generic SLM procedure is performed by using only the selected phase sequences for the original input sequence. Simulation results show that the proposed scheme significantly reduces the complexity in terms of IFFT and PAPR calculation than the conventional SLM, while maintaining the PAPR reduction performance.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.11C
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• pp.1076-1084
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• 2009

We propose an ATI (Adaptive Tone Injection) scheme based on clipping noise for PAPR (Peak-to-Average Power Ratio) reduction of OFDM (Orthogonal Frequency Division Multiplexing) signals. The proposed scheme is composed of three steps: clipping, tone selection, and TI procedures. In the first step, the peak samples in the IFFT (Inverse Fast Fourier Transform) outputs are scaled down by clipping. In the second step, the sub-carrier position where the power of the clipping noise is the maximum, is selected. Finally, the generic TI procedure is performed. Simulation results show that the proposed scheme does not require all the possible combinations of the original TI procedures, while maintaining the PAPR reduction performance.