• Journal of information and communication convergence engineering
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• v.9 no.1
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• pp.95-99
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• 2011

Digital forensics is the emerging research field for determining digital forgeries. Key issues of the tampered images are to solve the problems for detecting the interpolation factor and the tampered regions. This paper describes a method to detect the interpolation factors and the forged maps using the differential method and fast Fourier transform(FFT) along the horizontal, vertical, and diagonal direction, respectively from digital filtered tampered images. The detection map can be used to find out interpolated regions from the tempered image. Experimental results demonstrate the proposed algorithm proves effective on several filtering images by adobe $Photoshop^{TM}$ and show a ratio of detecting the interpolated regions and factors from digital filtered composite images.

• 제어로봇시스템학회:학술대회논문집
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• 2001.10a
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• pp.158.5-158
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• 2001

Monopulse active homing radar requires velocity and angle information of target to track fast moving target. Target velocity can be estimated by measuring the frequency shift between transmitted and received frequencies. Angle information is obtained by measuring boresight error. Measurement of doppler frequency component in received signal is done through FFT analysis and interpolation algorithm for fine tuning. Boresight errors in azimuth and elevation axes are proportional to the power of each difference channel relative to sum channel. The target signal power in difference channel is estimated more precisely by measuring the power of FFT result cell of maximum ...

• Proceedings of the IEEK Conference
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• 2003.07e
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• pp.2475-2478
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• 2003

CELP계열의 부호화기인 G.723.1 5.3kbps ACELP를 기반으로 하여 음질을 유지하면서 전송률을 낮출 수 있는 새로운 부호화 방법을 제안한다. 본 논문에서 적용한 부호화 방법은 음성 합성시 파라미터로 사용되는 지속시간 변경에 의해 CELP형 보코더의 전송률을 감소하고자 한다. 먼저 음성을 보코더 입력단에 입력하기 전 지속시간을 FFT 변환 특성을 이용해 음색의 변경 없이 지속시간을 줄임으써 계산시간을 줄이고 진폭과 위상 각각 1/2ⁿ배의 interpolation과 Decimation을 수행하여 부호화한다. 이렇게 부호화된 데이터는 G.723.1 복호화를 거치고, 다시 FFT point의 1/2ⁿ배 point로 IFFT과정을 수행함으로써 스팩트럼의 변경 없이 지속시간을 변경하여 원 음성을 합성하게 된다. G.723.1 보코더를 통과한 후 파형을 복원 실험한 결과 기존의 5.3kbps ACELP보다 46％정도 감소하였다.

• Proceedings of the Acoustical Society of Korea Conference
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• 1996.06a
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• pp.13-17
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• 1996

다수의 신호를 동시에 사용하는 경우에 나타나는 광대역 신호나 센서의 수와 형성하고자 하는 빔의 수가 많은 경우를 다루기 위해서는 주파수 영역 기법이 필요하다. FFT를 사용하는 낮은 샘플링 주파수에서도 시간 영역에서의 보간 방법보다 빠르게 정확한 시간 지연을 줄 수 있어 코히어런트 신호처리가 가능하다. 또한 특정 센서가 불량인 경우 보정이 상대적으로 용이하다는 장점을 가진다. 여러 가지 주파수 영역 빔형성 기법 중 계산량과 저장 용량면으로 효율적인 방법은 CZT와 FIR interpolation 방법이다. 또한, 공액 빔을 형성할 경우에는 Goertzel의 알고리듬을 이용하는 방법도 효율적이다.

• Wind and Structures
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• v.9 no.3
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• pp.231-243
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• 2006

An improvement to the spectral representation algorithm for the simulation of wind velocity fields on large scale structures is proposed in this paper. The method proposed by Deodatis (1996) serves as the basis of the improved algorithm. Firstly, an interpolation approximation is introduced to simplify the computation of the lower triangular matrix with the Cholesky decomposition of the cross-spectral density (CSD) matrix, since each element of the triangular matrix varies continuously with the wind spectra frequency. Fast Fourier Transform (FFT) technique is used to further enhance the efficiency of computation. Secondly, as an alternative spectral representation, the vectors of the triangular matrix in the Deodatis formula are replaced using an appropriate number of eigenvectors with the spectral decomposition of the CSD matrix. Lastly, a turbulent wind velocity field through a vertical plane on a long-span bridge (span-wise) is simulated to illustrate the proposed schemes. It is noted that the proposed schemes require less computer memory and are more efficiently simulated than that obtained using the existing traditional method. Furthermore, the reliability of the interpolation approximation in the simulation of wind velocity field is confirmed.

• Journal of the Korean Institute of Telematics and Electronics
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• v.25 no.4
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• pp.430-436
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• 1988

LSF(Line Spectral Frequency) derived from LPC has known as a very useful transmission parameter of speech signals, for it has a good linear interpolation characteristics and a low spectrum distortion at low bit rates coding. This paper presents that it is possible to extract directly the formant frequencies of speech signals from LSF parameter without application of FFT algorithm by comparing the distribution of LSF parameter with the frequency distribution of analysis filter. This paper suggests the advanced algorithm that results in improving the speed of convergence at analytic solution method. Also, for the flexibility of parameters, the process that transforms from LSF to LPC is presented.

• Journal of the Korean Institute of Telematics and Electronics D
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• v.34D no.5
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• pp.67-74
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• 1997

In this paper, we psresent a new scheme to synthetically generate a HPO digital hologram for a three-dimensional image that is modelled as the horizontally stacked two-dimensional images. The proposed method transforms a lightwave field into the angular spectrum of planewaves, which enables this method to use FFT routines, rather than using numerous arithmetic calculations. Hence, this method may be able to not only lead to the dramatically less computation but provide relatively excellent performances due to the phase error-free transformation. We present sampling constraints and implementaton procedure to obtian a hololine for each image and also point out the necessity of interpolation. Simulatioj results are presented to show the comparison with the conventional method in terms of computation time and performances, including the behaviors resulting form the different selection of parameter values to be used in the interpolations.

• Proceedings of the Acoustical Society of Korea Conference
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• autumn
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• pp.41-44
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• 2001

본 논문에서는 음성합성 과정에서 음성유닛을 연결할 때 모음의 결합부분에서 포만트의 불일치로 일어나는 부자연스러운 합성음이 발생되는 문제점을 개선하기 위해서 앞에 오는 음성 유닛과 뒤에 오는 합성 유닛의 포만트 변경에 관한 방법을 제안한다. 요즘에 연구되는 코퍼스 방식에선 에너지와 피치와 음순지속시간 등을 기준으로 유닛을 선택한 후 연결하지만, 스펙트럼의 불일치가 이루어진다. 이런 스펙트럼의 불일치는 음질의 저하를 유도한다. 그래서 앞 음성유닛의 연결부분의 일정부분과 뒤 음성 유닛의 연결부분의 일정부분의 포만트를 천이시켜 일치시켜줌으로써 음질을 향상시켰다. 음성신호를 FFT한 후 magnitude와 phase를 분리한 후 앞 음성의 연결부분의 magnitude와 뒷 음성의 연결부분의 magnitude를 기준으로 linear interpolation한 값을 목표치로 이동하고 다시 합하여 원 신호를 복원하는 방식으로 포만트를 변경시켰다.

• 제어로봇시스템학회:학술대회논문집
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• 2000.10a
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• pp.97-97
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• 2000

In this paper, a signal processing algorithm for software GPS receiver is proposed. The signal processor takes snapshot of the sampled If signal from the RF section of the GPS receiver. All the processing for code and carrier tracking and correlation are implemented using the digital signal processing techniques. In order to achieve fast code acquisition, correlation of the incoming GPS signal is performed using the FFT method, After code acquisition, to reduce the Doppler shift effect and increase the accuracy, the interpolation or the tracking are performed. The performance of the proposed processing algorithm is first evaluated using matlab/simulink. A signal acquisition board for sampling and logging GPS IF signal form the Mitel GPS RF chip set is constructed. In order to analyze the performance of the designed algorithm the experiments are performed and the results are analyzed.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.2
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• pp.37-45
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• 2007

A lot of conventional algorithms has tried to cancel the Residual Frame Synchronization Error (RFSE), which causes the performance degradation of channel estimation when using interpolation in lattice-type pilot-aided Time Division Duplex (TDD) OFDM systems. Among them, a pre- and post-compensation method has been attractive to eliminate the effects of RFSE, providing good performance but requiring high computational complexity. In this paper we propose a new method which lowers computational costs by adjusting the starting point of TDD OFDM systems' FFT window in order to compensate for the RFSE. For the performance verification, we apply the proposed method to IEEE 802.16e standard and also estimate additionally required computational power of it compared to the pre- and post-compensation method. Simulation and implementation results show that the proposed algorithm has performance similar to the conventional method and is very efficient with low power implementation, requiring only 4.9 % of the power which the conventional method needs additionally.