• 전기학회논문지
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• 제65권7호
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• pp.1247-1251
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• 2016

This study aims to investigate the effect of sampling frequency to the time domain and frequency domain analysis of pulse rate variability (PRV). Typical time domain variables - AVNN, SDNN, SDSD, RMSSD, NN50 count and pNN50 - and frequency domain variables - VLF, LF, HF, LF/HF, Total Power, nLF and nHF - were derived from 7 down-sampled (250 Hz, 100 Hz, 50 Hz, 25 Hz, 20 Hz, 15 Hz, 10 Hz) PRVs and compared with the result of heart rate variability of 10 kHz-sampled electrocardiogram. Result showed that every variable of time domain analysis of PRV was significant at 25 Hz or higher sampling frequency. Also, in frequency domain analysis, every variable of PRV was significant at 15 Hz or higher sampling frequency.

• Journal of electromagnetic engineering and science
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• 제8권1호
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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.

• 대한전자공학회:학술대회논문집
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• 대한전자공학회 2002년도 하계종합학술대회 논문집(5)
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• pp.9-12
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• 2002

The problem of sub-Nyquist nonuniform sampling for the perfect reconstruction of signals with time-varying spectral contents is studied. The signals are assumed to have a known instantaneous bandwidth in time-frequency domain. As the function of time, the nonuniform sampling pattern of a given signal, that is, the instantaneous sampling frequency is determined by the observation of instantaneous bandwidth based on time-frequency analysis. The proposed sampling pattern guarantees the perfect reconstruction of nonuniform sampled signals under Nyquist-sampling rate in average.

• 한국해양공학회지
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• 제24권1호
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• pp.166-171
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• 2010

In this paper, the front-end of a digital receiver with a 25 kHz carrier frequency, 5 kHz symbol rate, and any excess-bandwidth is designed using two basic facts. The first is known as the uniform sampling theorem, which states that the sampled sequence might not suffer from aliasing even if its sampling rate is lower than the Nyquist sampling rate if the analog signal is a bandpass one. The other fact is that if the sampling rate is 4 times the center frequency of the sampled sequence, the front-end processing complexity can be dramatically reduced due to the half of the sampled sequence to be multiplied by zero in the demixing process. Furthermore, the designed front-end is simplified by introducing sub-filters and sub-sampling sequences. The designed front-end is composed of an A/D converter, which takes samples of a bandpass filtered signal at a 20 kHz rate; a serial-to-parallel converter, which converts a sampled bandpass sequence to 4 parallel sub-sample sequences; 4 sub-filter blocks, which act as a frequency shifter and lowpass filter for a complex sequence; 4 synchronized switches; and 2 adders. The designed front-end dramatically reduces the computational complexity by more than 50% for frequency shifting and lowpass filtering operations since a conventional front-end requires a frequency shifting and two lowpass filtering operations to get one lowpass complex sample, while the proposed front-end requires only four filtering operation to get four lowpass complex samples, which is equivalent to one filtering operation for one sample.

• Journal of Communications and Networks
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• 제5권4호
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• pp.319-327
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• 2003

We consider the problem of low-sampling rate high-resolution channel estimation and timing for digital ultrawideband (UWB) receivers. We extend some of our recent results in sampling of certain classes of parametric non-bandlimited signals and develop a frequency domain method for channel estimation and synchronization in ultra-wideband systems, which uses sub-Nyquist uniform sampling and well-studied computational procedures. In particular, the proposed method can be used for identification of more realistic channel models, where different propagation paths undergo different frequency-selective fading. Moreover, we show that it is possible to obtain high-resolution estimates of all relevant channel parameters by sampling a received signal below the traditional Nyquist rate. Our approach leads to faster acquisition compared to current digital solutions, allows for slower A/D converters, and potentially reduces power consumption of digital UWB receivers significantly.

• 제어로봇시스템학회:학술대회논문집
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• 제어로봇시스템학회 2000년도 제15차 학술회의논문집
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• pp.360-360
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• 2000

In this paper, a new criterion f9r determining the sampling rate of digital conroller is proposed. This paper will introduce a method fur determining the appropriate sampling rate of digital controller which can be substituted with the given analog controller, using phase margin and gain cross over frequency, not rising time or bandwidth of the closed-loop system. This method also guarantees performance of the system. Without exact modeling functions of the plant, abstracting those functions, this paper can achieve stability and aimed performance of the system, and this paper proved it with proper modeling functions.

• 한국통신학회논문지
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• 제38A권7호
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• pp.545-551
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• 2013

MQASK 수신기에서 수신 심볼의 타이밍 동기에 사용되는 타이밍 복원 루프 Timing Error Detector(TED) 와 입력신호의 표본화율을 제어하는 VCO또는 NCO 및 루프 필터로 구성된다. 여기서 수신신호의 심볼율과 수신기의 표본화율의 시간 축에서의 위상차를 검출하는 TED는 심볼율과 표본화율의 주파수차가 클 경우 정상동작을 하지 못하는 단점이 있다. 본 논문에서는 PLL의 주파수 검출기와 같은 역할을 타이밍 복원 루프에서 수행하여 타이밍 복원 입력 신호의 주파수 차가 매우 큰 경우에도 타이밍 복원을 가능하게 할 수 있는 심볼율 변별기(Symbol Rate Discriminator SRD) 와 이를 사용한 타이밍 복원루프 구조를 제안 하였으며 이를 통해 심볼율이 가변되는 신호에 대한 타이밍 동기 획득이 가능함을 모의실험을 통해 입증하였다.

• 한국통신학회논문지
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• 제31권12C호
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• pp.1249-1256
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• 2006

본 논문은 대역통과 샘플링(bandpass sampling)이론을 사용하여 N개 RF 신호를 하나의 통신기기에서 직접 하향변환(down-conversion)할 때 요구되는 유효 샘플링 주파수 영역(valid sampling frequency range) 및 최소 샘플링 주파수를 찾는 방법을 제안하였다. 이 방식은 기존의 방식보다 효율적이며 복잡성이 낮은 방식을 새롭게 고안하여 N개의 신호에 대하여 수식을 유도하고 일반화 하였으며 컴퓨터 모의실험(simulation)을 통해 이 수식들을 증명하였다. 그리고 제안한 알고리즘은 샘플링 주파수를 결정할 때 인접 IF 신호간의 보호대역(guard-band)이 고려되어 실제적인 하드웨어 설계 측면에 도움이 되도록 하였다. 또한 기존의 방식과의 비교를 통해 그 복잡성(complexity) 및 정확성에서도 우수한 성능을 보여주었다.

• 디지털산업정보학회논문지
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• 제16권1호
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• pp.93-98
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• 2020

Research on underwater sensor networks is increasing due to such reasons as marine resource management, maritime disaster prediction and military protection. Many underwater sensor networks performs wireless communication using an acoustic sound wave band signal having a relatively low frequency. So the digital part of their modem can take charge of carrier band signal processing. To enable this, the sampling rate of the baseband band signal should be increased to a sampling rate at which carrier band signal processing is possible. In this paper, we designed a sampling rate increasing circuit based on a CIC interpolator for underwater sensor nodes. The CIC interpolator has a simple circuit structure. However, since the CIC interpolator has a large attenuation of the pass band and a wide transition band, an inverse sinc LPF is added to compensate for frequency response of the CIC interpolator. The proposed interpolator was verified in time domain and frequency domain using ModelSim and Matlab.

• 한국통신학회논문지
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• 제32권12A호
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• pp.1286-1295
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• 2007

본 논문은 SDR(software-defined radio) 시스템에 적용할 수 있는 기술로서, 대역통과 샘플링(bandpass sampling)이론을 사용하여 N개 RF 신호를 하나의 통신기기에서 직접 하향변환(down-conversion)할 때 요구되는 유효 샘플링 주파수 영역(valid sampling frequency range) 및 최소 샘플링 주파수를 찾는 방법을 제안하였다. 이 방식은 IF단의 신호배치에 대한 모든 순열을 고려하면서도 기존의 방식보다 효율적이며 복잡성이 낮은 방식을 새롭게 고안하여 N개의 신호에 대하여 수식을 유도하고 일반화 하였으며 컴퓨터 모의실험(simulation)을 통해 이 수식들을 증명하였다. 그리고 제안한 알고리즘은 샘플링 주파수를 결정할 때 인접 IF(intermediate frequency) 신호간의 보호대역(guard-band)이 고려되어 실제적인 하드웨어 설계 측면에 도움이 되도록 하였다. 또한 기존의 방식과의 비교를 통해 그 복잡성(complexity)및 정확성에서도 우수한 성능을 보여주었다.