• 한국음향학회지
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• 제36권4호
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• pp.238-246
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• 2017

고주파수, 고해상도 초음파 영상을 획득하기 위해서는 고주파수 초음파 변환기가 가지는 작은 구경에 따른 낮은 민감도와 침투 깊이에 따른 높은 감쇠량을 극복하여야 한다. 이는 고주파수 초음파 변환기의 본질적인 한계점이므로 수신 시스템을 통하여 그 한계점을 극복하여야 한다. 본 논문은 고주파수, 고해상도 초음파 영상을 위하여 고주파수 초음파 변환기의 본질적인 한계인 낮은 민감도와 높은 감쇠량을 극복할 수 있는 저잡음 광대역 수신 시스템 개발과 특성 평가 결과에 관한 것이다. 개발한 저잡음 광대역 수신 시스템은 80 MHz 이상의 동작 주파수 대역에서 최대 73 dB의 증폭 이득과 48 dB의 가변 이득 범위를 만족하며 ${\pm}1dB$의 증폭 이득 평탄도 성능을 가진다. 또한 개발한 수신 시스템은 상용 리시버에 비하여 8.4 dB 이상의 신호대잡음비 성능과 3.7 dB 이상의 대조도 성능이 우수함을 확인하였다.

• 대한전자공학회논문지SP
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• 제46권1호
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• pp.87-94
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• 2009

잡음예측 및 저감방법은 음성통신과 인식분야의 중요한 핵심기술이다. 본 논문에서는 다양한 잡음환경에 적용할 수 있는 새로운 잡음예측 및 저감 방법을 제안한다. 제안된 알고리즘은 시간 및 주파수영역의 noisy power spectrum 의 분산과 그 값의 정규화 ratio를 기반으로 한다. 제안한 방법은 다양한 잡음환경에서 잘 동작 할 수 있도록 적응추적 임계값을 사용하며, 이 임계값은 음성왜곡과 잔여잡음 사이의 trade-off를 제어한다. 새로운 알고리즘의 성능은 다양한 잡음환경에서 ITU-T P.835(SIG) and segment (SNR) 의해 평가하여 기존의 방법에 비해 향상된 결과를 나타냈다.

• Journal of electromagnetic engineering and science
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• 제17권1호
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• pp.39-43
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• 2017

In this paper, a systematic and statistical calibration technique was implemented to calibrate a high-speed signal converting system containing multiple digital-to-analog converters (DACs). The systematic error (especially the imbalance between DACs) in the current combining network of the multi-DAC system was modeled and corrected by calculating the path coefficients for individual DACs with wideband reference signals. Furthermore, by applying a Kalman filter to suppress noise from quantization and clock jitter, accurate coefficients with minimum noise were identified. For correcting an arbitrary waveform generator with two DACs, a co-simulation platform was implemented to estimate the system degradation and its corrected performance. Simulation results showed that after correction with 4.8 Gbps QAM signal, the signal-to-noise-ratio improved by approximately 4.5 dB and the error-vector-magnitude improved from 4.1% to 1.12% over 0.96 GHz bandwidth.

• 한국정보통신학회:학술대회논문집
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• 한국정보통신학회 2014년도 추계학술대회
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• pp.795-797
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• 2014

영상처리는 스마트폰, 태블릿 PC와 같은 영상 매체를 이용한 디스플레이 등의 발전으로 인해 여러 분야에서 필요성이 높아지고 있다. 디지털 영상 기술은 데이터를 획득, 처리, 전송하는 과정에서 여러 원인에 의해 잡음이 발생한다. 따라서 본 논문은 임펄스 잡음에 훼손된 잡음 영상을 복원하기 위해 방향성을 고려하여 잡음 제거에 우수한 메디안 필터를 제안하였으며, 영상내의 잡음 제거 특성을 입증하기 위해, 판단기준으로 PSNR(peak signal to noise ratio)을 사용하여 기존방법들과 비교하였다.

• International Journal of Computer Science & Network Security
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• 제24권3호
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• pp.182-188
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• 2024

Hardware components are an integral part of Hardware Define Radio (HDR) for seamless operations and optimal performance. On the other hand, Software Define Radio (SDR) is a program that does not rely on any hardware components for its performance. Both of the latter radio programmers utilize modulation functions to make their core components from signal processing viewpoint. The following paper concentrates on SDR based modulation and their performance under different modulations. The bit error rate (BER) of modulations such as PSK, QAM, and PSAM were used as indicators to test channel quality estimation in planar Rayleigh fading. Though it is not commonly used for channel fading, the method of the adder determines the regionally segmented channel fading. Thus, the estimation error of the channel change substantially reduces the performance of the signal, hence, proving to be an effective option. Moreover, this paper also elaborates that BER is calculated as a function of the sample size (signal length) with an average of 20 decibels. Consequently, the size of the results for different modulation schemes has been explored. The analytical results through derivations have been verified through computer simulation. The results focused on parameters of amplitude estimation error for 1dB reduction in the average signal-to-noise ratio, while the combined amplitude deviation estimation error results are obtained for a 3.5 dB reduction

• 전자공학회논문지B
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• 제31B권7호
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• pp.33-41
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• 1994

A single channel adaptive noise canceling (ANC) technique with a recursive time delay estimator (RTDE) is presented for removing effects of additive noise on the speech signal. While the conventional method makes a reference signal for the adaptive filter using the pitch estimated on a frame basis from the input speech, the proposed method makes the reference signal using the delay estimated recursively on a sample-by-sample basis. As the RTDEs, the recursion formulae of autocorrelation function (ACF) and average magnitude difference function (AMDF) are derived. The normalized least mean square (NLMS) and recursive least square (RLS) algorithms are applied for adaptation of filter coefficients. Experimental results with noisy speech demonstrate that the proposed method improves the perceived speech quality as well as the signal-to-noise ratio and cepstral distance when compared with the conventional method.

• 한국군사과학기술학회지
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• 제4권1호
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• pp.147-154
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• 2001

The underwater explosion shock test is performed for the evaluation of the shock-resistant capability which is a very critical factor considering the survivability of the battle ship. Some measured signals have impulsive noise and gaussian white noise because of the unstable power supply system and the transient movement of cables during the underwater explosion shock test. The advanced shock signal analysis method which remove the noise of measured signal using the threshold policy of the median filter and the orthogonal wavelet coefficients are proposed. It is verified that the signal-to-noise ratio was improved about 30㏈ by the numerical simulation.

• 전자공학회논문지A
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• 제33A권6호
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• pp.32-41
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• 1996

We propose the improved GMSK (gaussian-filtered minimum shift keying) quadrature modulator using the FIR(finite impulse response )filter whose coefficients are obtained form the differnce of phase response, and design its ASIC (applicaton specific integrated circuit) which can be used for GSM (global system for mobile communication) digital cellular system and DCS 1800 (digital cellular system at 1800MHz) personal communication system. Input data become quantized I and Q channel 10 bit signal through cosine and sine ROM mapping after being filtered by the FIR filter whose normalized bandwidth is 0.3 and designed by considering intersymbol interference as well as sampling ratio. These two signals become the GMSK modulated I and Q channel signal through DAC (digital-to-analog converter) and 7th order analog chebyshev LPF(low pass filter) respectively. The difference between the ideal analog signal and its digitized signal is analyzed in terms of sampling noise, quantization noise, truncation noise and coefficient noise. And the effect of the LPF following the DAC is considered. The ASIC design of the GMSK quadrature modulator is also confirmed by an experiment.

• 디지털콘텐츠학회 논문지
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• 제11권3호
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• pp.283-290
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• 2010

1.5T에서 경직장 코일과 3.0T의 위상 배열 코일을 이용하여 신호대 잡음비를 비교하여 3.0T의 위상 배열 코일의 유용성을 제시한다. 전립선암으로 진단받은 환자 40명을 대상으로 1.5T MR scanner에서는 경직장 표면 코일과 3.0T MR scanner에서는 위상 배열 코일을 이용하여 T2 강조 영상, T1 강조영상, 확산 강조 영상을 각각 획득 후 전립선의 Central Zone(CZ)과 Peripheral Zone(PZ)에서 Right Peripheral Zone(PZ)과 Left Peripheral Zone(PZ) 세 부위에서 신호 강도(signal intensity)측정을 하였다. 정량적 분석방법으로 관심영역의 신호대 잡음비(signal to noise ratio)를 구하고 평균화 하여 1.5T MR scanner로 획득한 그룹과 3.0T MR scanner로 획득한 그룹에서의 신호대 잡음비를 비교하였다. 3.0T MR scanner에서 위상 배열 코일을 사용한 경우 1.5T MR scanner에서 경직장 코일을 사용한 경우보다 신호대 잡음비가 증가됨을 알 수 있었다.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2000년도 춘계학술대회논문집
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• pp.1710-1715
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• 2000

Estimation of damping ratio for vibration signals measured on the passenger car's seat is useful for the objective evaluation of impact harshness in car. The vibration signal is a transient signal represented by many coupled modes of suspension system. Wavelet transform automatically decouples these modes in the time-frequency domain. Damping ratios for decoupled modes are obtained by logarithmic treatment for the Wavelet transformed signal. The objective evaluation using Wavelet transform has been well corresponded with subjective evaluation done by skilled engineers.