• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.2C
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• pp.191-199
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• 2006

In this paper, we propose a method to implement a predistorter of the $rho$-th order inverse filter structure to prevent signal distortion and spectral re-growth due to the high PAPR (peak-to-average ratio) of the OFDM signals and the non-linearity of high-power amplifiers. We model the memory-less non-linearity of the high-power amplifier with a polynomial model and utilize the inverse of the model, the $rho$-th order inverse filter, for the predistorter. Once the non-linearity is modeled with a polynomial, since we can determine the $rho$-th order inverse filter only with the coefficients of the polynomial, large memory is not required. To update the coefficients of the non-linear high-power amplifier model, we can use LMS or RLS algorithms. The convergence speed is high since the number of coefficients is small, and the computation is simple since manipulation of complex numbers is not necessary.

• The Journal of the Acoustical Society of Korea
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• v.23 no.4
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• pp.333-339
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• 2004

This Paper proposes the vector quantizer-pyramid vector quantizer(VQ-PVQ) structure. Also both predictive structure and safety-net concept are combined into the VQ-PVQ to quantize the IPC parameter of wideband speech codec. The Performance is compared to the LPC vector quantizer used in the AMR-WB(ITU-T G.722.2). demonstrating reduction in both spectral distortion and encoding memory.

• ETRI Journal
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• v.23 no.1
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• pp.16-22
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• 2001

Efficient quantization methods of the line spectrum pairs (LSP) which have good performances, low complexity and memory are proposed. The adaptive quantization range method utilizing the ordering property of LSP parameters is used in a scalar quantizer and a vector-scalar hybrid quantizer. As the maximum quantization range of each LSP parameter is varied adaptively on the quantized value of the previous order's LSP parameter, efficient quantization methods can be obtained. The proposed scalar quantization algorithm needs 31 bits/frame, which is 3 bits less per frame than in the conventional scalar quantization method with interframe prediction to maintain the transparent quality of speech. The improved vector-scalar quantizer achieves an average spectral distortion of 1 dB using 26 bits/frame. The performances of proposed quantization methods are also evaluated in the transmission errors.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.60 no.5
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• pp.1023-1029
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• 2011

In the industrial vision systems, image correction has great influence on the overall performance of measurement or inspection. The overall area of distorted image is usually splitted into small control areas, and each area is corrected by its control points. The performance of correction methods using control points can be improved by reduction of control areas because the computational time and memory highly depend on the number of control areas. We develop a merging algorithm that reduces control areas and preserves the correction accuracy. The algorithm merges the splitted control areas by use of quad tree method. Experimental results are presented to verify the usefulness of the proposed method.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.34S no.4
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• pp.50-56
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• 1997

In this paper, efficient quantization methods of line spectrum pairs (LSP) which has good performances and low complexity and memory are proosed for vocoder of mobile communication system. The adaptive quantization method utilizing the ordering property of LSP parameters is used in a scalar quantizer and a vector-scalar hybrid quantizer. The proposed scalar quantization algorithm needs 31 bits/frame to maintain the transparent quality of speech. The improved vector-scalar quantizer achieves an average spectral distortion of 1dB using 26 bits/frame. The proposed methods are evaluated in the channel errors and changed the predictor structure to maintain the robustness to channel errors.

• Proceedings of the KIEE Conference
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• 1990.07a
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• pp.11-14
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• 1990

The Complex Harmonic Balance Finite Element Method CHBFEM ) is dicussed for the time - periodic magnetic field with saturation characteristics. And Jw - method which is used for analyzing liner system with sinusoidal voltage input can be generalized in nonlinear time-periodic magnetio field system. The CHBFEM enables us to calculate the each harmonic magnetic flux ditribution and the distortion of currents resulting from material at an AC voltage source and to save calculating time, the number of calculation and computer memory.

• 한국정보디스플레이학회:학술대회논문집
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• 2005.07a
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• pp.276-279
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• 2005

The proposed Novel OverDriving Technology with Optimum Look-up-Table（LUT）is suggested to be a better solution to reducing the occurrence of overshooting caused by the traditional overdriving method implemented to normally liquid crystal (VA mode). Chunghwa Picture Tubes, LTD. (CPT) has successfully implement this tech into 20 inch TFTLCD TV module at the present day. The proposed technology can speed up gray-to-gray response time of LCD less than one half of frame time. On the side, Optimum LUT construction apply the compression scheme to record total actual grayscale transfer characteristics instead of applying the normal spread method such as linear / non-linear interpolation. The memory space is been reduced and the distortion of the image quality is lesser.

• Journal of electromagnetic engineering and science
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• v.8 no.1
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• pp.17-22
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• 2008

It is presented that sampling rates for behavioral modeling of quasi-memory less nonlinear devices can be far less than the Nyquist rate of the input signal. Although it has been believed that the sampling rate of nonlinear device modeling should be at least the Nyquist rate of the output signal, this paper suggests that far less than the Nyquist rate of the input signal can be applied to the modeling of quasi-memoryless nonlinear devices, such as frequency multipliers. To verify, a QPSK signal at 820 MHz were applied to a frequency tripler, whereby the device can be utilized as an up-converting mixer into 2.46 GHz with the aid of digital predistortion. AM-AM, AM-PM and PM-PM can be successfully measured regardless of sampling rates.

• Proceedings of the Acoustical Society of Korea Conference
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• 1994.06a
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• pp.1098-1103
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• 1994

In this paper, we design and implement the Korean speech synthesis by rule system. This system is applied the multiband excitation signal on voiced sounds. The multiband excitation signal is obtained by mixing impluse spectrum and which noise spectrum. We find that the quality of synthesized speech is improved using this application. Also, we classify the voiced sounds by cepstral euclidian distance measure for reducing overhead memory. The representative excitation signal of the same group's voiced sounds is used as excitation signal on synthesis. This method does not affect the quality of synthesized speech. As the result of experiment, this method eliminates the "buzziness" of synthesized speech and reduces the spectral distortion of synthesized speech.ed speech.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.05a
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• pp.673-676
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• 2015

Digital Pre-Distortion (DPD) is a linearization technique for nonlinear power amplifiers (PAs) by implementing inverse function of the PA at baseband digital stage. To obtain proper DPD parameters, a feedback path is required to convert the PA output to a baseband signal, and a memory is also needed to store the feedback signals. DPD parameters are usually found by an adaptive algorithm from the feedback samples. However, for the adaptive algorithm to converge to a reliable solution, long feedback samples are required, which increases convergence time and hardware complexity. In this paper, we propose a DPD technique that requires relatively short feedback samples. From the observation that the convergence time of the adaptive algorithm highly depends on the initial condition, this paper iteratively utilizes the feedback samples while keeping and using the converged DPD parameters at the former iteration as the initial condition at the current iteration. Computer simulation results show that the proposed method performs better than the conventional technique while the former requires much shorter feedback samples than the latter.