• The Journal of the Convergence on Culture Technology
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• v.6 no.3
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• pp.427-430
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• 2020

The multi-channel analog signal to digital signal conversion is increasing in the field of IoT and medical measurement equipments. It has chip area and power consumption constraints to use a few single or 2_channel ADC for multi_channel application. This paper described to design and implement a proposed comb filter for multi-channel, 24bit ADC. The function of proposed comb filter is verified by matlab simulation and the FPGA test board. It was fabricated using SK Hynix 0.35㎛ CMOS standard process. The performance and chip size is compared with the existing design method that uses integrator/differentiator and FIR construction. The proposed comb filter is expected to use the IoT product and medical measurement equipments that require multi-channel, low power consumption and small hardware size.

• Proceedings of the Acoustical Society of Korea Conference
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• 1998.06e
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• pp.3.2-6
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• 1998

본 논문에서는 실측된 실내 충격 응답을 모델링하여 실내 잔향 필터 설계를 시도하였다. 급강하법(steepest descent method)을 이용하여 측정된 실내 충격 응답을 4개의 콤 필터(comb filter)와 2개의 올패스 필터(allpass filter)로 이루어진 잔향 필터로 모델링하여, 잔향 필터의 계수를 결정하였다.

• Journal of Korea Multimedia Society
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• v.9 no.5
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• pp.542-553
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• 2006

Because Y and C signals in a composite video signal are piled one on another in the same frequency, it is impossible to separate them completely. Therefore, it is necessary to develop efficient separation technique in order to minimize degradation of video quality. In this paper, we propose wavelet-based 3D comb filter algorithm and architecture for separating Y and C signals from a composite video signal. The proposed algorithm uses wavelet transform and thresholding of compared lines for acquiring the maximum video quality. Simulation results show that the proposed algorithm has better image quality and better PSNR than previous algorithms. For real application of the proposed algorithm, we developed a hardware architecture and the architecture was implemented by using VHDL. Finally, a VLSI layout of the proposed architecture was generated by using 0.25 micrometer CMOS process.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.16 no.8
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• pp.1753-1758
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• 2012

This paper introduces analog FIR filters based on a distributed amplifier and analyzes the proposed filter's characteristics. A simple design method of an analog FIR filter based on the digital filter design technique is also introduced. The proposed analog FIR filters are a moving average(MA) and a comb type filters with no multiplier. This simple structures of the proposed filters may enable to operate at multi-GHz frequency range and applicable to combine a filter and an amplifier of RF system. The proposed analog FIR filters were implemented with standard $0.18{\mu}m$ CMOS technology. The designed GHz analog FIR filters are simulated by Cadence Spectre and compared to the results of digital FIR filters obtained from MATLAB simulations. From the simulation results, the characteristics of the proposed analog FIR filters are fairly well matched with those of digital FIR filters.

• Proceedings of the Korea Electromagnetic Engineering Society Conference
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• 2003.11a
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• pp.288-293
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• 2003

In this paper, Filter could use to IMT2000 repeater using resonator of Comb-line form that has waveguide structure and designed. Coupled resonators designed using slot in wall between resonators to compact. 5-stage resonators have equal structure and input and output part was optimized to get into minimum impedance. Band pass filter characteristic satisfied design condition as result that used Chebyshev function that have 0.2 dB ripple and analyze using simulation tool.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2016.11a
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• pp.163-164
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• 2016

이전까지 많은 연구자들은 적응 신호처리(Adaptive Signal Process)를 이용한 잡음 제거 방법을 연구해 왔다. 그러나, 최근 발전하고 있는 멀티콥터는 프로펠러 모터의 RPM(Revolution Per Minute)이 실시간으로 변하기 때문에 적응 신호처리를 이용하여도 깔끔한 결과를 얻어 내기가 어렵다는 한계가 존재한다. 또한, 특정 주파수를 기준으로 형성되는 고조파(Harmonics)는 적응 알고리즘인 (N)LMS 를 이용한 예측에서 오차를 발생시키는 문제를 발생시킨다. 따라서, 본 논문에서는 멀티콥터를 이용한 음향 취득에 대한 소음 저감 방법으로 회전 속도계(Tachometer), 콤 필터(Comb Filter), NLMS 알고리즘(Normalized Least Mean Square Algorithm)을 이용한 방법을 제안한다.

• Journal of Broadcast Engineering
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• v.3 no.1
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• pp.1-12
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• 1998

8VSB transmission system of American standard and OFDM system of European standard are to be international standards for the transmission of digital terrestrial television. When countries using PAL standard adopt the American 8VSB system which is designed under the circumstance of NTSC system, one of strong interference is the co-channel PAL interference coming from analog PAL signal. In this paper, in order to solve this problem, we classified seven types of PAL system into three groups according to the bandwidth and a new PAL-8VSB system is proposed to reject the co-channel PAL interference by using modified comb filters suitable for each group. By computer simulations of the proposed PAL-8VSB system with the co-channel PAL interference it is confirmed that the proposed comb filter results in the improvement of DIU ratio by about 9 dB.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.7 no.4
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• pp.156-160
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• 1982

A Comb Filter(C.F.) is constucted with a N-stages one-dimensional B.B.D.(Bucket-Brigade Device) delay line. One channel of the B.P.F. (Band Pass Filter) Bank is experimented, which includes a R.F.(Recursie Filter) using S/H circuits cascaded to the C.F. This algorithm of the C.F.B.(Comb Filter Bank) becomes the parallel spectrum analyzer circuit. The algorithm has less number of multiplication than that of FFT and improves the SNR.

• The Journal of the Acoustical Society of Korea
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• v.25 no.8
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• pp.362-369
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• 2006

In this paper, we improve a Physical modeling of sanio gayageum and synthesize sounds by right-hand playing styles. Parameters of loop filter are estimated from decay region of recorded sound. That results in improved accuracy and reduction of computational costs. Body is implemented to a resonator which has characteristics of main resonances extracted from impulse response. Residual signal with main resonances removed is used as excitation signal of proposed model. Amok alter is approximated to frequency response of amok and is implemented to the 15th order all-role digital filter. Beating (by middle and index finger) among the right-hand playing styles is represented by feedforward comb filter. Parameters of this filter are extracted from recorded sound. The synthesized sounds using improved physical model of sanjo gayageum. plucking and beating, are pretty similar to original sounds.

• MALSORI
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• no.46
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• pp.65-76
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• 2003

In this paper, we employ the adaptive comb filtering for effective noise reduction in mobile communication environment. Adaptive comb filtering is a well-known method for noise reduction, but requires correct pitch period and must be applied just in voiced speech frames. To satisfy these requirements we use two kinds of information extracted from speech packets, one of which is the pitch period information measured precisely by a speech coder and the other is the frame rate information related to a decision on speech or silence frame. Experiments on speech recognition system confirm the efficiency of this method. Feature parameters employing this method give superior performance in noise environment to those extracted directly from output speech.