• 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.

• 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.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.43 no.5 s.311
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• pp.48-56
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• 2006

In this paper, we proposed the high quality audio watermarking algorithm using MDCT/IMDCT (Modified DCT/Inverse Modified DCT) with psychoacoustic model. Generally, a digital audio watermark is embedding the frequency domain after frequency transform of the digital audio data but the digital audio quality is affected by watermarking. In our scheme, the digital audio data is spread with PN((Pseudo Noise) code and then audio watermark is embedded in MDCT processing that refers psychoacoustic model. In MDCT processing, according to the shape of filter bank output, the block switching selects a window sequence that has 256, 1,024 or 2,048 points interval for high quality audio. The author confirm that when watermark weight ${\alpha}$ is 2.5 below, the detection ratio of watermark is a satisfied to SDMI's(Secure Digital Music Initiative) recommendation 50% above and SM is $50{\sim}68dB$ range with mainly 4 kind of attacks(Compression, Cropping, FFT and Echo).

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38A no.12
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• pp.1030-1038
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• 2013

The proposed Alamouti coded OFDM effectively cancels Inter Carrier Interference (ICI) due to frequency offset between distributed antennas. The conventional Alamouti coded OFDM schemes to mitigate ICI utilize N-point Inverse Fast Fourier Transform/Fast Fourier Transform (IFFT/FFT) operations for OFDM modulation and demodulation processes with total N subcarriers. However, the performance degrades because ICI is also repeated in N periods due to the property of N-point IFFT/FFT operation. In order to avoid this problem, null data are used at the subcarriers with large ICI and thus, data rate decreases. The proposed scheme employs 2N-point IFFT/FFT instead of N-point IFFT/FFT in order to increase sampling rate. By increasing sampling rate, the amount of interference significantly decreases because the period of ICI also increases. The proposed scheme increases the data rate and improves the performance by reducing amount of ICI and the number of null-data. Furthermore, the gain of the performance and data rate of the proposed scheme is significant with higher modulation such as 16-Quadarature Amplitude Modulation (QAM) or 64-QAM.

• Journal of the Korean Institute of Telematics and Electronics
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• v.26 no.10
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• pp.1617-1623
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• 1989

The introduction of the fast Fourier transform(FFT),an efficient computational algorithm for the discrete Fourier transform(DFT) by Cooley and Tukey(1965), has brought to the limelight various other discrete transforms. Some of the analog functions from which these transforms have been derived date back to the early 1920's, for example, Walsh functions (Walsh, 1923) and Hadamard Transform(Enomoto et al, 1965). Fast algorithms developed for the forward transform are equally applicable, exept for minor changes, to the inverse transform. In this paper, we present a simple pipelined Hadamard matrix(HM) which is used to develop a fast algorithm for the Hadamard Processor (HP). The Fast Hadamard Transform(FHT) can be derived using matrix partitioning techniques. The HP system is incorporated through a modular design which permits tailoring to meet a wide range of video data link applications. Emphasis has been placed on a low cost, a low power design suitable for airbone system and video codec.

• Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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• 2000.06a
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• pp.1939-1944
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• 2000

Different test methods were used to measure and compare the loss factor of a steel beam. They are free vibration logarithmic decay method, half-power bandwidth method using narrow-band analysis, wide-band analysis method with inverse FFT. In these tests, specimens are clamped at one end. Free vibration method and half-power bandwidth method gave good results. Effect of vibration pick-ups were tested. We also tried center excitation method but could not obtain loss factor.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.12 no.3
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• pp.285-291
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• 1987

Seismic waves are attenuated by losses of energy as they propagate through the earth. One way to model this numerically is to make the velocity a complex number, the real part giving the phase velocity and the imaginary part the attenuation. This models wave propagation in a medium for which the logarithmic decrement is independent of frequency(attenuation coefficient is proportional to frequncy). The aim is to modify forward and inverse numerical modeling so that attenuation can be specified as a function of position.

• Journal of the Korea Institute of Military Science and Technology
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• v.7 no.2 s.17
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• pp.88-99
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• 2004

A development of a software on radar target signature analysis is presented in this paper The target signature includes Radar Cross Section(RCS) prediction, Range Profile(RP) processing and Inverse Synthetic Aperture Radar(ISAR) processing. Physical Optics(PO) is the basic calculation method for RCS prediction and Geometrical Optics(GO) is used for ray tracing in the field calculation of multiple reflection. For RP and ISAR, Fast Fourier Transform(FFT) and Matrix Pencil(MP) method were implemented for post-processing. Those results are integrated into two separate softwares named as Radar Target Signature Generator(RTSG) and Radar Target Signature Analyser(RTSA). Several test results show good performances in radar signature prediction and analysis.

• Proceedings of the KSMRM Conference
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• 2002.11a
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• pp.75-75
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• 2002

목적： 고자장 자기공명영상 시스템 등으로 신호대잡음비가 향상됨에 따라 데이터 측정에서 analog-to-digital converter (ADC)의 quantization noise 가 중요한 시스템 사양으로 부각되고 있다. 특히 자기공명영상은 공간주파수 영역에서 데이터를 측정하기 때문에 dc와 ac간의 신호 차이가 매우 크며, 이러한 dynmic range는 3-D 영상에서 더욱 커진다. 본 연구에서는 다양한 자기총명 영상기법 및 실험 파라미터에 따른 신호의 dynamic range와 ADC의 bit 수에 따른 quantization noise를 살펴봄으로써, 주어진 시스템에 적합한 ADC의 bit 수를 분석하고자 한다. 대상 및 방법： 펄스 시퀀스의 종류, 파라미터, 2D/3D 등에 따른 각 신호의 크기를 수학적으로 모델링하여 신호의 크기를 예측하였다. 또한 whole body MRI 시스템에서 실험을 통하여 신호의 크기를 비교하였다. ADC의 quantization noise를 실험과 시뮬레이션을 통하여 살펴보았다. 시뮬레이션은 test 영상을 Inverse FFT 하여 spatial frequency domain data를 만든 후, 다양한 bit 수의 ADC로 quantization을 한 후 다시 영상을 재구성하였다. 재구성된 영상과 원영상 간의 error가 quantization noise가 된다. 또한 이러한 error가 주파수 영역에서의 error 값과 일치하는지를 확인하였다.

• Journal of the Korean Society for Precision Engineering
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• v.22 no.5 s.170
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• pp.72-79
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• 2005

Gearing system emits inconsistent noise from gear teeth impact in case of gear defects. But, it is not easy for inspection operator in production line to distinguish objectively the defective product. Therefore, customer complains continuously bad noise of the geared motor. Because impulsive signal at low frequency has a tendency not to appear in frequency domain, it is difficult to separate the gear inconsistent noise of defective gear from overall geared motor's noise using general signal processing method such as FFT. In this paper, the method to estimate more objectively the inconsistent noise of gearing system and to measure the quantities is suggested. Suggested method uses Cepstrum, Autocorrelation, Comb Lifter and Inverse Cepstrum by turns to make objective quantities about noise level.