• The Journal of the Acoustical Society of Korea
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• v.23 no.8
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• pp.554-561
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• 2004

As the technologies of semiconductor and multimedia communication have been improved. the high-quality video and the multi-channel audio have been highlighted. MPEG Audio Layer 3 decoder has been implemented as a Processor using a standard. Since the synthesis filter of MPEG-1 Audio Layer 3 decoder requires the most outstanding operation in the entire decoder. the synthesis filter that can reduce the amount of operation is needed for the design of the high-speed processor. Therefore, in this paper, the synthesis filter. the most important part of MPEG Audio, is materialized in FPGA using the method of DAULT (distributed arithemetic look-up table). For the design of high-speed synthesis filter, the DAULT method is used instead of a multiplier and a Pipeline structure is used. The Performance improvement by 30% is obtained by additionally making the result of multiplication of data with cosine function into the table. All hardware design of this Paper are described using VHDL (VHIC Hardware Description Language) Active-HDL 6.1 of ALDEC is used for VHDL simulation and Synplify Pro 7.2V is used for Model-sim and synthesis. The corresponding library is materialized by XC4013E and XC4020EX. XC4052XL of XILINX and XACT M1.4 is used for P&R tool. The materialized processor operates from 20MHz to 70MHz.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.1B
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• pp.89-98
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• 2000

This paper presents an active noise cancelling model using wavelet transform and subband filter banks based on adaptive filter. The analysis filter banks decompose input and error signals into QMF filter banks of lowpass and highpass bands. Each filter bank uses wavelet filter with dyadic tree structure. The decomposed input and error signals are iterated by adaptive filter coefficients of each subband using filtered-X LMS algorithm. The synthesis filter banks make output signal of wideband with perfect reconstruction to prepare adaptive filter output signals of each subband. The analysis and synthesis niter hants use conjugate quadrature filters for Pefect reconstruction. Also, The delayed LMS algorithm model for on-line identification of error path transfer characteristics is used gain and acoustic time delay factors. The proposed adaptive active noise cancelling modelis suggested by system retaining the computational and convergence speed advantage using wavelet subband filter banks.

• 한국신재생에너지학회:학술대회논문집
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• 2008.10a
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• pp.246-249
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• 2008

IGCC (석탄가스화 복합발전) 시스템 공정 중 가스화기에서 발생되는 합성가스 내에는 높은 농도의 분진들이 함유되어 있기 때문에 하부 공정들을 보호하기 위해서는 가스화기 후단에 집진장치가 필히 설치되어야 한다. 집진장치의 설계 제작 운전에 있어서 점착성 분진층 부착,분진 브리징, 필터 파손 현상 등과 같은 다양한 문제점들로 인하여 고온의 합성가스를 처리하는 데는 아직 기술적 한계들이 산재해 있는 상황이다. 이와 같은 문제점들은 필터에 부착된 분진층의 탈진성능을 향상시키거나, 필터의 열피로를 줄일 수 있도록 필터 재생을 위한 탈진의 횟수를 줄이는 방법을 통하여 해결될 수 있다. 본 연구에서는 파일롯 규모의 IGCC용 세라믹캔들필터 집진장치에서의 가스 유입 및 탈진 조건에 따른 집진장치의 차압 특성 변화를 수치해석과 실험을 통하여 분석함으로써 탈진성능을 향상시킬 수 있는 조건과 탈진 횟수를 최소화시킬 수 있는 방안을 도출하고자 하였다.

• Journal of the Institute of Convergence Signal Processing
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• v.23 no.3
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• pp.179-184
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• 2022

In this paper, we are going to study how we can automatically determine the number of convolutional filters for the optimal model without a search algorithm. This paper proposes HSE Block by connecting SE Block proposed in SENet to a convolutional neural network and connecting a convolutional neural network not learned at the bottom. An experiment was conducted to increase the number of filters by one per 3 epoch using two datasets for the HSEBlock model and to increase the number of filters by the value in the filter. Based on this experiment, the model was constructed with multi-layer HSE Block instead of layer HSE Block, and the experiment was carried out using a dataset that was more difficult to learn than the one used in the previous experiment. The effect of HSE Block was verified by conducting an experiment with the number of HSE Blocks set to 2, 3, 4, and 5 on a dataset that is more difficult to learn than before.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.13 no.6
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• pp.510-520
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• 1988

In this paper, the output error variance due to the differential vcariation of the state-space coefficient [ABCD], which is the coefficient quentization error, is normalized on the variance for cases that infinite wordlength state-space digital filter is realized by the finite one. That is, defining S as the statistical sensitivity and extending controllability gramian, observability gramian, and 2nd order mode analysis method to the state space digital filter, we synthesize the realization structure with the minimum statistical sensitivity and prove the effecency of the minimum statistical sensitivity structure synthesis by the simulation.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.1
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• pp.9-14
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• 2012

In this study, the diplexer used a separator or a combine of signals was designed and its characteristics were investigated. The diplexer consists of two fixed tuned bandpass filter such as low pass filter(LPF) and high pass filter(HPF) sharing a common part and was simulated. The diplexer with micro-strip structure was fabricated. The LPF of diplexer has insertion loss of -0.4 dB, return loss of -30 dB and 1.6 GHz cut-off frequency and HPF has insertion loss of -0.8 dB, return loss of -30 dB and 1.9 GHz cut-off frequency, respectively.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2006.06a
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• pp.37-40
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• 2006

FM(Frequency Modulation) 음 합성방식은 오랫동안 연구되어 왔고 여러 가지 효과를 주기에 유용하고 비교적 낮은 사양으로 높은 수준의 배음을 합성할 수 있지만 그 응용분야가 대부분이 서양악기 위주의 연구였다. 이에 본 논문에서는 국악 관악기인 피리를 주파수 영역에서 저주파 영역과 고주파 영역으로 나누어 분석하고 이 결과를 이용하여 FM 파라미터를 추출하고 비대칭 FM합성방식으로 1차적인 소리를 합성하였다. 분석된 저주파 영역과 고주파 영역의 특성을 각각의 역 필터링을 위한 2개의 필터를 설계하여 이를 다시 합성된 소리를 통과시켜 두 소리를 합산함으로써 더욱 원음과 유사한 소리를 합성할 수 있었다.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2020.11a
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• pp.46-48
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• 2020

HEVC (High Efficiency Video Coding)의 In-Loop 필터 중 하나인 디블로킹 필터는 예측과 변환 블록 주변의 균일하지 않은 잡음을 제거하기 위해 사용된다. 영상을 복원할 때 텍스쳐 정보의 경우, 디블로킹 필터로 눈에 보이는 잡음을 제거하는 데 유리하지만 깊이 정보는 깊이를 예측하여 합성하는 데 사용되므로 디블로킹 필터를 적용하면 합성에 방해가 될 수 있다. 이에 본 논문은 TMIV (Test Model 6 for MPEG Immersive Video) 인코더를 거쳐 나온 기본 시점과 추가 시점의 텍스쳐와 깊이 영상에 디블로킹 필터를 적용하여 BD-Rate의 향상 정도와 인지 화질적 관점에서의 화질 개선 여부를 실험을 통해 검증한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.6C
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• pp.874-882
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• 2004

MPEG Audio compression algorithm is the international standard for the digital compression of high quality audio using mechanism of the perceptual coding based on psychoacoustic masking. It is necessary to discuss the constraints on designing of common filter banks for MPEG-2 BC and MPEG-2 AAC decoder system, which is not Down yet, mapping audio signals from the time domain into the frequency domain. In this paper, we present an architecture of common synthesis filter whcih can be used for MPEG-2 BC and MPEG-2 AAC decoder using recursive structure. The proposed algorithm is based on recursive architecture that effectively performs common compulsion.

• The Journal of Korean Institute of Electromagnetic Engineering and Science
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• v.16 no.1 s.92
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• pp.19-25
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• 2005

This paper describes a synthesis method for a self-equalized dual-passband filter. Compared to conventional dual-passband filter, a self-equalized dual-passband filter can reduce BER(bit error rate) in digital data communications and does not need an external equalizer for group-delay equalization. To validate the synthesis technique, a 10-pole dual-mode dual-passband filter which has two self-equalized 5-pole elliptic response passband is synthesized.