• ETRI Journal
• /
• 제32권2호
• /
• pp.214-221
• /
• 2010

Radio frequency (RF) subsampling can be used by radio receivers to directly down-convert and digitize RF signals. A goal of a cognitive radio/software defined ratio (CR/SDR) receiver design is to place the analog-to-digital converter (ADC) as near the antenna as possible. Based on this, a band-pass sampling (BPS) frontend for CR/SDR is proposed and verified. We present a receiver architecture based second-order BPS and signal processing techniques for a digital RF frontend. This paper is focused on the benefits of the second-order BPS architecture in spectrum sensing over a wide frequency band range and in multiband receiving without modification of the RF hardware. Methods to manipulate the spectra are described, and reconstruction filter designs are provided. On the basis of this concept, second-order BPS frontends for CR/SDR systems are designed and verified using a hardware platform.

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

• 음성과학
• /
• 제10권2호
• /
• pp.249-258
• /
• 2003

This study attempted to examine a speaker identification method using difference sum and correlation coefficient determined from a pair of intensity level matrices of band-pass-filtered numeric sounds produced by ten female speakers of similar age and height. Subjects recorded three digit numbers at a quiet room at a sampling rate of 22 kHz on a personal computer. Collected data were band-pass-filtered at five different band ranges. Then, matrices of five intensity levels at 100 proportional time points were obtained. Pearson correlation coefficients and the sum of absolute intensity differences between a pair of given matrices were determined within and across the speakers. Results showed that very high correlation coefficient and small difference sum generally occurred within each speaker but some individual variation was also observed. Thus, the matrix pair with a higher coefficient and a smaller difference sum was averaged to form each individual's model. Comparison among the speakers yielded generally low coefficients and large differences, which suggests successful speaker identification, but among them there were a few cases with very high coefficients and small differences. Future studies will focus on finer band ranges and additional spectral parameters at some peak points of the intensity contour at a low frequency band.

• 한국감성과학회:학술대회논문집
• /
• 한국감성과학회 2002년도 춘계학술대회 논문집
• /
• pp.44-50
• /
• 2002

The fast Fourier transform (FFT) is a good method to estimate spectral density, but the frequency resolution is limited to the sampling window, and thus the precise characteristics of the spectral density for short signals are not clear. To solve the limitation, a multiple band-pass filter was introduced to estimate the precise time course of the spectral density for flash visual evoked potentials (VEPs). Signals were recorded during -200 and 600 ms using balanced noncephalic electrodes, and sampled at 1 K Hz in 12 bits. With 1 Hz and 10 ms resolutions, spectral density was estimated between 10 and 100 Hz. Background powers at the alpha-and beta-bands were high over the posterior scalp, and powers around 200ms were evoked at the same bands over the same region, corresponding to P110 and N165 of VEPs. normalized's spectral density showed evoked powers around 200 ms and suppressed powers following the evoked powers over the posterior scalp. The evoked powers above the 20Hz band were not statistically significant. However, the gamma band was significantly evoked intra-individually; details in the gamma bands were varied among the subjects. Details of spectral density were complicated even for a simple task such as watching flashes; both synchronization and desynchronization occurred with different distributions and different time courses.

• 한국항해항만학회:학술대회논문집
• /
• 한국항해항만학회 2006년도 International Symposium on GPS/GNSS Vol.2
• /
• pp.385-390
• /
• 2006

This paper examines the sampling and jitter specifications and considerations for Global Navigation Satellite Systems (GNSS) software receivers. Software radio (SWR) technologies are being used in the implementation of communication receivers in general and GNSS receivers in particular. With the advent of new GPS signals, and a range of new Galileo and GLONASS signals soon becoming available, GNSS is an application where SWR and software-defined radio (SDR) are likely to have an impact. The sampling process is critical for SWR receivers, where it occurs as close to the antenna as possible. One way to achieve this is by BandPass Sampling (BPS), which is an undersampling technique that exploits aliasing to perform downconversion. BPS enables removal of the IF stage in the radio receiver. The sampling frequency is a very important factor since it influences both receiver performance and implementation efficiency. However, the design of BPS can result in degradation of Signal-to-Noise Ratio (SNR) due to the out-of-band noise being aliased. Important to the specification of both the ADC and its clocking Phase- Locked Loop (PLL) is jitter. Contributing to the system jitter are the aperture jitter of the sample-and-hold switch at the input of ADC and the sampling-clock jitter. Aperture jitter effects have usually been modeled as additive noise, based on a sinusoidal input signal, and limits the achievable Signal-to-Noise Ratio (SNR). Jitter in the sampled signal has several sources: phase noise in the Voltage-Controlled Oscillator (VCO) within the sampling PLL, jitter introduced by variations in the period of the frequency divider used in the sampling PLL and cross-talk from the lock line running parallel to signal lines. Jitter in the sampling process directly acts to degrade the noise floor and selectivity of receiver. Choosing an appropriate VCO for a SWR system is not as simple as finding one with right oscillator frequency. Similarly, it is important to specify the right jitter performance for the ADC. In this paper, the allowable sampling frequencies are calculated and analyzed for the multiple frequency BPS software radio GNSS receivers. The SNR degradation due to jitter in a BPSK system is calculated and required jitter standard deviation allowable for each GNSS band of interest is evaluated. Furthermore, in this paper we have investigated the sources of jitter and a basic jitter budget is calculated that could assist in the design of multiple frequency SWR GNSS receivers. We examine different ADCs and PLLs available in the market and compare known performance with the calculated budget. The results obtained are therefore directly applicable to SWR GNSS receiver design.

• 한국시뮬레이션학회:학술대회논문집
• /
• 한국시뮬레이션학회 2001년도 The Seoul International Simulation Conference
• /
• pp.313-317
• /
• 2001

The approach to predict time series without neglecting the fluctuation in a short period is tried by using a digital FIR filter and a neural network. The differential waveform of the Nikkei average closing price is filtered by the FIR band-pass filter of 101 length. It is filtered into the five frequency bands of 0-1Hz, 1-2Hz, 2-3Hz, 3-4Hz and 4-5Hz by setting the sampling frequency 10Hz. The each filtered waveform is learned and forecasted by the neural network. The neural network of the back propagation method is adopted in the learning the waveform. By inputting the data of 20 days in the past, the prediction of 10 days ahead is carried out. After learning the time series of each frequency band by the neural network, the predicted data far each frequency band are obtained. The predicted waveforms of each frequency band are synthesized to obtain a final forecast. The waveform can be forecasted well as a whole.

• 한국음향학회지
• /
• 제14권5호
• /
• pp.89-93
• /
• 1995

균일표본화에서 나타나는 샘플간의 잉여정보를 더욱 줄임으로써, 요구되는 데이타량을 크게 줄일 수 있는 방법으로 비균일표본화 방법이 고려된다 그러나, 음성신호의 경우 이러한 비균일표본화 방법을 바로 적용하면, 필요한 데이타량이 균일표본화에 견주어 크게 줄어들지 않게 된다. 특히, 잡음환경하에서는 오히려 균일표본화의 경우보다도 데이타량이 커질 수 있다. 이러한 단점을 보완하기 위해서, 먼저 음성신호를 적당히 저대역 필터링을 한 후 비균일표본화를 적용하고, 고대역성분에서의 오차는 잡음신호로 보완하는 방법을 제안한다. 제안된 방법은 기존의 비균일표본화 방법보다 약1.8배의 데이타압축효과를 얻을 수 있었다.

• 전력전자학회논문지
• /
• 제18권3호
• /
• pp.225-231
• /
• 2013

This paper presents an improved signal processing technique in the square-wave-type voltage injection method for IPMSM sensorless drives. Since the sensorless method based on the square-wave voltage injection does not use low-pass filters to get an error signal for estimating rotor position and allows the frequency of the injected voltage signal to be high, the sensorless drive system may achieve an enhanced control bandwidth and reduced acoustic noise. However, this sensorless method still requires low-pass and band-pass filters to extract the fundamental component current and the injected frequency component current from the motor current, respectively. In this paper, these filters are replaced by simple arithmetic operations so that the time delay for estimating the rotor position can be effectively reduced to only one current sampling. Hence, the proposed technique can simplify its whole signal process for the IPMSM sensorless control using the square-wave-type voltage injection. The proposed technique is verified by the experiment on the 800W IPMSM drive system.

• 한국통신학회논문지
• /
• 제40권1호
• /
• pp.9-15
• /
• 2015

S급 전력 증폭기 응용을 위한 CMOS 대역 통과 델타 시그마 변조기(BPDSM)와 캐스코드 E급 전력 증폭기를 설계 및 제작 하였다. 대역 통과 델타 시그마 변조기는 1 GHz의 샘플링 주파수로 250 MHz의 입력 신호를 펄스폭 변조 방식의 디지털 신호로 변조하며 양자화 잡음을 효과적으로 제거하였다. 대역 통과 델타 시그마 변조기는 25 dB의 SQNR을 가지며 1.2 V 전원 전압에서 24 mW의 전력을 소비한다. 캐스코드 E급 전력 3.3V 전원에서 동작하며 최대 18.1 dBm의 출력 전력을 가지며 25%의 드레인 효율을 보였다. 두 회로 모두 동부 0.11 um RF CMOS 공정으로 제작되었다.

• 제어로봇시스템학회:학술대회논문집
• /
• 제어로봇시스템학회 1994년도 Proceedings of the Korea Automatic Control Conference, 9th (KACC) ; Taejeon, Korea; 17-20 Oct. 1994
• /
• pp.682-688
• /
• 1994

LSM(Least-Squares Method) has inherent limitation that precise system identification over wide frequency band is difficult especially at low frequency hand. In this paper we propose to use decimation, a spectrum analysis method widely used in signal processing. The merits of decimation are the flexibility of selection of the frequency hand concerned and the function of LPF(Low Pass Filter). In this paper, frequency-domain is divided into separate frequency bands which will be combined into full frequency-domain by using MDM(Multiple Decimation Method). In this way, free selection of sampling frequency for each hand is possible and the low frequency oscillation modes of LSM are avoided.