• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.12A
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• pp.1177-1181
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• 2006

In this paper, we proposed the new wavelength tunable filter for wide tunability of fiber ring laser. The proposed filter consists of the cascaded connection of Fiber Reflection Mach-Zehnder Interferometer and Sagnac filter or of hybrid filter and fiber Reflection Mach-Zehnder Interferometer. The simulation shows that the continuous tunable range of 50nm can be obtained and that more stable single mode operation of ring laser can be expected due to narrow FWHM. of filter spectrum.

• Proceedings of the Korea Information Processing Society Conference
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• 2001.10b
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• pp.1279-1282
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• 2001

최근 인터넷의 급속한 사용 증가로 인해 전송 링크의 광대역 지원과 멀티미디어 트래픽의 QoS 보장 문제, 향상된 IP 서비스의 제공 문제 해결은 필수적이다. MPLS 기술은 IP의 유연성과 확장성을 제공할 수 있는 패러다임의 하나이다. MPLS 망의 경계에 위치하는 LER 은 링크 계층 뿐만 아니라 IP 계층에서도 패킷을 전달해야 한다. 본 논문에서는 하드웨어 포워딩 엔진을 갖는 MPLS LER에서, 패킷 필터링 기능을 지원하기 위해서 포워딩 엔진을 제어하기 위한 기법에 대해서 기술한다. 이러한 패킷 필터링 기능은 MPLS LER에 firewall 기능을 제공하는데 응용될 수 있다.

• The Journal of the Acoustical Society of Korea
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• v.20 no.1
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• pp.13-20
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• 2001

Adaptive digital filters with long impulse response such as acoustic echo canceller and active noise controller suffer from slow convergence and computational burden. Subband techniques and multirate signal processing have been recently developed to improve the problem of computational complexity and slow convergence in conventional adaptive filter. Any FIR transfer function can be realized as a serial connection of interpolators followed by subfilters with a sparse impulse response. In this case, each interpolator which is related to the column vector of Hadamard matrix has band-pass magnitude response characteristics shifted uniformly. Subband technique using Hadamard transform and decimation of subband signal to reduce sampling rate are adapted to system modeling and acoustic noise cancellation In this paper, delayless subband structure with nonuniform bandwidth has been proposed to improve the performance of the convergence speed without aliasing due to decimation, where input signal is split into subband one using tree-structured filter bank, and the subband signal is decimated by a decimator to reduce the sampling rate in each channel, then subfilter with sparse impulse response is transformed to full band adaptive filter coefficient using Hadamard transform. It is shown by computer simulations that the proposed method can be adapted to general adaptive filtering.

• The Journal of the Acoustical Society of Korea
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• v.23 no.7
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• pp.521-531
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• 2004

In this paper we developed the Physical modeling of the Sanio Gayageum using the improved digital waveguide theory. The frequency characteristics of the Gayageum body is implemented by an inverse filtering and the impulse response of the body. We obtained the synthesis sounds of the unit sound for the Gayageum using the simulation of the straight-line fits by the changes of the fundamental frequencies depending on the Amok location. Finally. we could obtain the virtual Sanio Gayageum sounds similar to the actual Gayageum by tuning the Amok positions.

• Journal of the Institute of Convergence Signal Processing
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• v.18 no.1
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• pp.13-20
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• 2017

In this paper, we evaluate the basic test results of the 32 Gbps observational equipment introduced as the back-end system for the wideband VLBI (Very Long Baseline Interferometry) observation of KVN (Korean VLBI Network). Radio astronomers want to make a large radio telescope that has excellent performance in order to observe the superfine structure of a celestial body, but a lot of money is needed. Therefore, in order to increase the sensitivity, the performance improvement of the receiving system and the method of observing the wide frequency bandwidth are introduced. To do this, we adopted a wideband sampling method for converting analog signals to digital with ultra-fast speeds and a wideband sampler for performing digital filtering in order to observe a wide observational frequency bandwidth. The wideband sampler (OCTAD-K) supports up to 16 Gsps-2bits sampling and supports a variety of observational bandwidth using digital filtering techniques. In particular, it is designed to support KVN's 4-frequency simultaneous observation system and VERA(VLBI Exploration of Radio Astrometry)'s 2-beam observation system. It can also support polKVN(Korean VLBI Network), KaVA(KVN and VERA Array), 32Gbps Direct Sampler, Digital Filter, Widebandarization observations and supports the standard VDIF(VLBI Data Interchange Format) format of observed data. In this paper, the performance of the system and the problem solving are described in detail after performing the factory inspection and field test before the system is introduced.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.11C
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• pp.1098-1110
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• 2003

In this paper, we propose a homomorphic wavelet-based MMSE filter with edge detection to restore images degraded by speckle noise. In the proposed method, a noisy image is first transformed into logarithmic domain. Each pixel in the transformed image is then classified into flat and edge regions by applying DIP operator to the image restored by homomorphic directional MMSE filter. Each pixel in flat region is restored by homomorphic wavelet-based MMSE filter. Each pixel in edge region is restored by the weighted sum of the output of homomorphic wavelet-based MMSE filtering and that of homomorphic directional MMSE filtering. The restored image in spatial domain is finally obtained by applying the exponential function to the restored image in logarithmic domain. Experimental results show that the restored images by the proposed method have ISNR improvement of 3.3-4.0 ㏈ and ${\beta}$, a measurement parameter on edge preservation, improvement of 0.0103-0.0126 and superior subjective image quality over those by conventional methods.

• Proceedings of the IEEK Conference
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• 2001.09a
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• pp.809-812
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• 2001

본 논문에서는 AMR(Adaptive Multi-Rate)과 MLT (Modulated Lapped Transform) 벡터 양자화 방법을 이용하여 광대역 음성부호화기를 설계하였다. 제안한 음성부호화 알고리즘은 split-band 구조를 가지고 있으며 16kHz로 샘플링 된 신호를 입력받아 QMF 필터에 의해 두 개의 대역으로 나누어, 각각 8kHz 샘플링 신호로 변환시킨 후 저대역(0Hz-3400Hz)의 신호와 고대역(3400Hz -7000Hz)의 신호로 나누어 각각 부호화한다. 나누어진 두 개의 협대역 음성신호는 AMR(Adaptive Multi-Rate)부호화기와 MLT (Modulated Lapped Transform)벡터 양자화 방법을 사용하여 각각 부호화되어 전송된다. 수신단에서는 각 대역을 AMR과 IMLT(Inverse MLT) 벡터 양자화 방법으로 역부호화하여 음성신호를 합성한다. 제안한 음성부호화기는 20.2kbps에서 12.15kbps까지의 다전송률로 동작된다. 설계된 광대역 음성부호화기는 MOS시험 결과로부터 G.722의 56 kbps 음성이 설계된 코더의 20.2 kbps와 비슷한 음질을 갖음을 확인할 수 있었다.

• Proceedings of the IEEK Conference
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• 2000.09a
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• pp.755-758
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• 2000

본 논문에서는 AMR(Adaptive Multi-Rate)를 이용하여 광대역 음성부호화기를 설계하였다. 16kHz로 샘플링 된 입력 신호를 QMF 필터에 의해 두 개의 대역으로 나누어, 각각 decimation하여 두 개의 8kHz 샘플링 신호로 변환시킨 후 저대역(0Hz-3400Hz)의 신호와 고대역(3400Hz -7000Hz)의 신호로 나누어 각각 부호화한다. 나누어진 두 개의 협대역 음성신호는 AMR(Adaptive Multi-Rate)과 ATC(Adaptive Transform Coding)을 사용하여 각각 부호화되어 전송된다. 두 대역으로부터 부호화된 정보는 20.2kbps에서 12.75kbps까지의 전송률을 갖고, 수신단에서는 각 대역을 AMR과ATC방법으로 역부호화하여 음성신호를 합성한다. 설계된 광대역 음성부호화기의 성능을 평가하기 위해 ITU-T의 표준안인 G.722를 포함하여 MOS 시험을 하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.26 no.5B
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• pp.632-638
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• 2001

본 논문에서는 AMR(Adaptive Multi-Rate)를 이용하여 광대역 음성부호화기를 설계하였다. 16kHz로 샘플링된 입력 신호를 QMF 필터에 의해 두 개의 대역으로 나누어, 각각 decimation하여 두 개의 8kHz 샘플링 신호로 변환시킨 후 저대역(0Hz-3400Hz)의 신호와 고대역(3400Hz∼7000Hz)의 신호로 나누어 각각 부호화한다. 나누어진 두 개의 협대역 음성신호는 AMR(Adaptive Multi-Rate)과 ATC(Adaptive Transform Coding)을 사용하여 각각 부호화되어 전송된다. 두 대역으로부터 부호화된 정보는 20.2kbps에서 12.75kbps까지의 전송률을 갖고, 수신단에서는 각 대역을 AMR과 ATC 방법으로 역부호화하여 음성신호를 합성한다. 설계된 광대역 음성부호화기의 성능을 평가하기 위해 ITU-T의 표준안인 G.722를 포함하여 MOS 시험을 하였다.

• Journal of Broadcast Engineering
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• v.24 no.2
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• pp.217-226
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• 2019

In this paper, we propose a filtering method that can improve the results of inverse tone-mapping using guided image filter. Inverse tone-mapping techniques have been proposed that convert LDR images to HDR. Recently, many algorithms have been studied to convert single LDR images into HDR images using CNN. Among them, there exists an algorithm for restoring pixel information using CNN which learned to restore saturated region. The algorithm does not suppress the noise in the non-saturation region and cannot restore the detail in the saturated region. The proposed algorithm suppresses the noise in the non-saturated region and restores the detail of the saturated region using a WGIF in the input image, and then applies it to the CNN to improve the quality of the final image. The proposed algorithm shows a higher quantitative image quality index than the existing algorithms when the HDR quantitative image quality index was measured.