• Title/Summary/Keyword: 주파수 추정기

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Performance Analysis of Frequency Diversity Scheme for OFDM Systems Using Sub-channel Correlation Characteristics (부채별 상관 특성을 이용한 OFDM 시스템의 주파수 다이버시티 기법 성능 분석)

  • 이종식;김장욱;오창헌;조성준
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.29 no.6A
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    • pp.614-622
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    • 2004
  • In this paper, we propose the frequency diversity scheme for performance improvement of a OFDM system without decreasing the spectral efficiency. In the proposed scheme, information bit is encoded to symbol by a simple procedure, and the encoded symbol is transmitted through the two lowest correlated sub-channels with the particular phase difference. At the receiver, a frequency diversity gain is obtained by a simple signal processing. We also suggest optimum phase difference value to minimize the performance degradation which resulted from a phase difference estimation error and bit/symbol mapping method to minimize BER. As results, at the point of performance improvement, the proposed scheme is excellent even though it requires a little increase of system complexity because of an additional encoding and decoding. In particular, we confirmed through computer simulation that on the same channel environment and bandwidth efficiency, the 27x/1Rx STBC-OFDM system adopting the proposed frequency diversity scheme outperforms the conventional 27x/1Rx STBC-OFDM system performance

Adaptive Beamforming System Based on Combined Array Antenna (혼합 배열 안테나 기반의 적응 빔형성 시스템)

  • Kim, Tae-Yun;Hwang, Suk-Seung
    • The Journal of the Korea institute of electronic communication sciences
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    • v.16 no.1
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    • pp.9-18
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    • 2021
  • The 5G communication system employs the millimeter wave with the extremely high frequency. Since the high frequency signal has the strong straightness, the beamforming technology based on the multiple base stations is required for services covering wide range. The beamformer needs the angle-of-arrival(AOA) information of the signal incident to the antenna, and it is generally estimated through the high resolution AOA estimation algorithm such as Multiple Signal Classification (MUSIC) or Estimation of Signal Parameters via Rotational Invariacne Technique (ESPRIT). Although various antenna array shapes can be employed for the beamformer, a single shape (square, circle, or hexagonal) is typically utilized. In this paper, we introduce a transmitting/receiving beamforming system based on the combined array antenna with square and circular shapes, which is proper to various frequency signals, and evaluate its performance. For evaluating the performance of the proposed beamforming system based on the combined array antenna, we implement the computer simulation employing various scenarios.

Extraction of Cole-Cole Parameters from Time-domain Induced Polarization Data (시간영역 유도분극 자료로부터 Cole-Cole 변수 산출)

  • Kim, Yeon-Jung;Cho, In-Ky
    • Geophysics and Geophysical Exploration
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    • v.24 no.4
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    • pp.164-170
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    • 2021
  • Frequency-domain and time-domain induced polarization methods can provide spectral information about subsurface media. Analysis of spectral characteristics has been studied mainly in the frequency-domain, however, time-domain induced polarization research has recently become popular. In this study, assuming a homogeneous half-space model, an inversion method was developed to extract Cole-Cole parameters from the measured secondary potential or electrical resistivity. Since the Cole-Cole parameters of chargeability, time constant, and frequency index are not independent of each other, various problems, such as slow convergence rate, initial model problem, local minimum problem, and divergence, frequently occur when conventional nonlinear inversion is applied. In this study, we developed an effective inversion method using the initial model close to the true model by introducing a grid search method. Finally, the validity of the developed inversion method was verified using inversion experiments.

Sound Synthesis of Right-Hand Playing Styles Using Improved Physical Modeling of Sanjo Gayageum (개선된 산조 가야금의 물리적 모델링을 이용한 오른손 주법의 음 합성)

  • Cho, Sang-Jin;Chong, Ui-Pil
    • 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.

Speech Enhancement Based on Improved Minima Controlled Recursive Averaging Incorporating GSAP (전역 음성 부재 확률 기반의 향상된 최소값 제어 재귀평균기법을 이용한 음성 향상 기법)

  • Song, Ji-Hyun;Bang, Dong-Hyeouck;Lee, Sang-Min
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.49 no.1
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    • pp.104-111
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    • 2012
  • In this paper, we propose a novel method to improve the performance of the improved minima controlled recursive averaging (IMCRA). From an examination for various noise environment, it is shown that the IMCRA has a fundamental drawback for the noise power estimate at the offset region of continuity speech signals. Espectially, it is difficult to obtain the robust estimates of the noise power in non-stationary noisy environments that is rapidly changed the spectral characteristics such as babble noise. To overcome the drawback, we apply the global speech absence probability (GSAP) conditioned on both a priori SNR and a posteriori SNR to the speech detection algorithm of IMCRA. With the performance criteria of the ITU-T P.862 perceptual evaluation of speech quality (PESQ) and a composite measure test, we show that the proposed algorithm yields better results compared to the conventional IMCRA-based scheme under various noise environments. In particular, in the case of babble 5 dB, the proposed method produced a remarkable improvement compared to the IMCRA ( PESQ = 0.026, composite measure = 0.029 ).

Meaurement Algorithms for EDGE Terminal Performance Test (EDGE 단말기 성능 테스트를 위한 측정 알고리즘)

  • Kang, Sung-Jin;Hong, Dae-Ki;Kim, Nam-Yong
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.12
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    • pp.2719-2730
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    • 2009
  • In this paper, we implement the measurement functionality for performance measurements of EDGE (Enhanced Data Rates for GSM Evolution) terminal by using software. Generally speaking, the receiving algorithms in normal MODEM cannot be used directly to a measurement system due to the lack of accuracy. Therefore, we propose a new receiver algorithm for precise EDGE signal measurements. In the proposed algorithm, 2-stage (coarse stage, fine stage) parameters estimation (symbol-timing, frequency offset, carrier phase) scheme is used. To improve the estimation accuracy, we increase the number of the received signal samples by interpolation. The proposed EDGE signal measurement algorithm can be used for verifying the hardware measurement system, and also can be used for the commercial systems through software optimization.

Analysis of the Linear Amplifier/ADC Interface in a Digital Microwave Receiver (디지털 마이크로파 수신기에서의 선형 증폭기와 ADC 접속 해석)

  • Lee, Min Hyouck;Kim, Sung Gon;Choi, Hee Joo;Byon, Kun Sik
    • Journal of Advanced Navigation Technology
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    • v.3 no.1
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    • pp.52-59
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    • 1999
  • Digital microwave wideband receiver including linear amplifier, analog-to-digital converter(ADC) and digital signal processor is able to analyze its performance using sensitivity and dynamic range of system. Determination of gain, third-order intermodulation products and ADC characteristics and design criteria for the linear amplifier chain is essential problem for sensitive and dynamic range. Also, if there are two signals with frequencies very close, digital signal processor must be able to separate the two signals. In this paper, we measured dynamic range as gain was changed and determined gain value for the proper sensitivity and dynamic range and high resolution spectrum estimation was used to separate two close signals.

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Hardware Implementation of Context Modeler in HEVC CABAC Decoder (HEVC CABAC 복호기의 문맥 모델러 설계)

  • Kim, Sohyun;Kim, Doohwan;Lee, Seongsoo
    • Journal of IKEEE
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    • v.21 no.3
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    • pp.280-283
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    • 2017
  • HEVC (high efficiency video coding) exploits CABAC (context-based adaptive binary arithmetic coding) for entropy coding, where a context model estimates the probability for each syntax element. In this paper, a context modeler was designed and implemented for CABAC decoding. lookup table was used to reduce computation and to increase speed. 12 simulations for HEVC standard test sequences and encoder configurations were performed, and the context modeler was verified to perform correction operations. The designed context modeler was synthesized in 0.18um technology. Maximum frequency, maximum throughput, and gate count are 200 MHz, 200 Mbin/s, and 29,268 gates, respectively.

Montgomery Multiplier Supporting Dual-Field Modular Multiplication (듀얼 필드 모듈러 곱셈을 지원하는 몽고메리 곱셈기)

  • Kim, Dong-Seong;Shin, Kyung-Wook
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.24 no.6
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    • pp.736-743
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    • 2020
  • Modular multiplication is one of the most important arithmetic operations in public-key cryptography such as elliptic curve cryptography (ECC) and RSA, and the performance of modular multiplier is a key factor influencing the performance of public-key cryptographic hardware. An efficient hardware implementation of word-based Montgomery modular multiplication algorithm is described in this paper. Our modular multiplier was designed to support eleven field sizes for prime field GF(p) and binary field GF(2k) as defined by SEC2 standard for ECC, making it suitable for lightweight hardware implementations of ECC processors. The proposed architecture employs pipeline scheme between the partial product generation and addition operation and the modular reduction operation to reduce the clock cycles required to compute modular multiplication by 50%. The hardware operation of our modular multiplier was demonstrated by FPGA verification. When synthesized with a 65-nm CMOS cell library, it was realized with 33,635 gate equivalents, and the maximum operating clock frequency was estimated at 147 MHz.

Influences and Compensation of Phase Noise and IQ Imbalance in Multiband DFT-S OFDM System for the Spectrum Aggregation (스펙트럼 집성을 위한 멀티 밴드 DFT-S OFDM 시스템에서 직교 불균형과 위상 잡음의 영향 분석 및 보상)

  • Ryu, Sang-Burm;Ryu, Heung-Gyoon;Choi, Jin-Kyu;Kim, Jin-Up
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.21 no.11
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    • pp.1275-1284
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    • 2010
  • 100 MHz bandwidth and 1 Gbit/s data speed are needed in LTE-advanced for the next generation mobile communication system. Therefore, spectrum aggregation method has been studied recently to extend usable frequency bands. Also bandwidth utilization is increased since vacant frequencies are used to communicate. However, transceiver structure requires the digital RF and SDR. Therefore, frequency synthesizer and PA must operate over wide-bandwidth and RF impairments also increases in transceiver. Uplink of LTE advanced uses DFT-S OFDM using plural power amplifier. The effect of ICI increases in frequency domain of receiver due to phase noise and IQ imbalance. In this paper, we analyze influences of ICI in frequency domain of receiver considering phase noise and IQ imbalance in multiband system. Also, we separate phase noise and IQ imbalance effect from channel response in frequency domain of uplink system. And we propose a method to estimate the channel exactly and to compensate IQ imbalance and phase noise. Simulation result shows that the proposed method achieves the 2 dB performance gain of BER=$10^{-4}$.