• Title/Summary/Keyword: High Level Signal

Search Result 795, Processing Time 0.035 seconds

An artificial noise generation method for MODEM performance test in satellite communication system (위성통신 시스템에서 수신기 모뎀 성능을 시험하기 위한 인위 잡음 발생 방법)

  • Cho, Tae-Chong
    • The Journal of the Institute of Internet, Broadcasting and Communication
    • /
    • v.20 no.4
    • /
    • pp.59-64
    • /
    • 2020
  • Occasionally, MODEM test in satellite communication systems are needed. But Rx terminals in satellite communications are designed to obtain high SNR generally, therefore artificial bad conditions and environments are demanded for the test. One of the typical method is satellite output power reduction. Using noise generator can be another method. However, these costs a lot of money, time, and procedures in reality. In order to overcome these problems, this paper proposes an artificial noise generation method for MODEM test in satellite communication systems. First of all, SNR of a general heterodyne Rx terminal is calculated. Based on the calculation, a new model which is including variable attenuator is proposed to increase noise level. Simulation results illustrate the variable attenuator can control SNR, and these show that MODEM test in satellite communication systems be possible.

A Parallel Equalization Algorithm with Weighted Updating by Two Error Estimation Functions (두 오차 추정 함수에 의해 가중 갱신되는 병렬 등화 알고리즘)

  • Oh, Kil-Nam
    • Journal of the Institute of Electronics Engineers of Korea TC
    • /
    • v.49 no.7
    • /
    • pp.32-38
    • /
    • 2012
  • In this paper, to eliminate intersymbol interference of the received signal due to multipath propagation, a parallel equalization algorithm using two error estimation functions is proposed. In the proposed algorithm, multilevel two-dimensional signals are considered as equivalent binary signals, then error signals are estimated using the sigmoid nonlinearity effective at the initial phase equalization and threshold nonlinearity with high steady-state performance. The two errors are scaled by a weight depending on the relative accuracy of the two error estimations, then two filters are updated differentially. As a result, the combined output of two filters was to be the optimum value, fast convergence at initial stage of equalization and low steady-state error level were achieved at the same time thanks to the combining effect of two operation modes smoothly. Usefulness of the proposed algorithm was verified and compared with the conventional method through computer simulations.

Performance Evaluation of a Time- and Frequency-Domain Clipping-Based PAPR Reduction Scheme in a DVB-T System (DVB-T 시스템에서 시간 및 주파수 영역 클리핑 기반의 PAPR 감소기법의 성능평가)

  • Seo, Man-Jung;Im, Sung-Bin;Kim, Na-Hoon;Cho, Jun-Kyung
    • Journal of the Institute of Electronics Engineers of Korea TC
    • /
    • v.44 no.1
    • /
    • pp.24-31
    • /
    • 2007
  • DVB-T (Digital Video Broadcasting-Terrestrial) is an important multimedia broadcasting technology capable of high data-rate transmission and adopted by Europe. OFDM (Orthogonal Frequency Division Multiplexing) is the backbone technique employed in DVB-T to support multimedia services that have various bandwidths. Unfortunately, an OFDM signal has a large PAPR (Peak-to-Average Power Ratio). In this paper, we investigate the performance of a simple PAPR reduction scheme for the DVB-T system, which requires no change of a receiver structure or no additional information transmission. The approach we employed is clipping in the time and frequency domains. The time-domain clipping is carried out with a predetermined clipping level while the frequency-domain clipping is done within EVM (Error Vector Magnitude). This approach is suboptimal with lower computational complexity compared to the optimal method. The simulation results demonstrate that the proposed one is getting more effective at lower modulation levels and with more allowed constellation error.

2.6 GHz GaN-HEMT Power Amplifier MMIC for LTE Small-Cell Applications

  • Lim, Wonseob;Lee, Hwiseob;Kang, Hyunuk;Lee, Wooseok;Lee, Kang-Yoon;Hwang, Keum Cheol;Yang, Youngoo;Park, Cheon-Seok
    • JSTS:Journal of Semiconductor Technology and Science
    • /
    • v.16 no.3
    • /
    • pp.339-345
    • /
    • 2016
  • This paper presents a two-stage power amplifier MMIC using a $0.4{\mu}m$ GaN-HEMT process. The two-stage structure provides high gain and compact circuit size using an integrated inter-stage matching network. The size and loss of the inter-stage matching network can be reduced by including bond wires as part of the matching network. The two-stage power amplifier MMIC was fabricated with a chip size of $2.0{\times}1.9mm^2$ and was mounted on a $4{\times}4$ QFN carrier for evaluation. Using a downlink LTE signal with a PAPR of 6.5 dB and a channel bandwidth of 10 MHz for the 2.6 GHz band, the power amplifier MMIC exhibited a gain of 30 dB, a drain efficiency of 32%, and an ACLR of -31.4 dBc at an average output power of 36 dBm. Using two power amplifier MMICs for the carrier and peaking amplifiers, a Doherty power amplifier was designed and implemented. At a 6 dB back-off output power level of 39 dBm, a gain of 24.7 dB and a drain efficiency of 43.5% were achieved.

Realistic Keyboard Typing Motion Generation Based on Physics Simulation (물리 시뮬레이션에 기반한 사실적인 키보드 타이핑 모션 생성)

  • Jang, Yongho;Eom, Haegwang;Noh, Junyong
    • Journal of the Korea Computer Graphics Society
    • /
    • v.21 no.5
    • /
    • pp.29-36
    • /
    • 2015
  • Human fingers are essential parts of the body that perform complex and detailed motion. Expression of natural finger motion is one of the most important issues in character animation research. Especially, keyboard typing animation is hard to create through the existing animation pipeline because the keyboard typing typically requires a high level of dexterous motion that involves the movement of various joints in a natural way. In this paper, we suggest a method for the generation of realistic keyboard typing motion based on physics simulation. To generate typing motion properly using physics-based simulation, the hand and the keyboard models should be positioned in an allowed range of simulation space, and the typing has to occur at a precise key location according to the input signal. Based on the observation, we incorporate natural tendency that accompanies actual keyboard typing. For example, we found out that the positions of the hands and fingers always assume the default pose, and the idle fingers tend to minimize their motion. We handle these various constraints in one solver to achieve the results of real-time natural keyboard typing simulation. These results can be employed in various animation and virtual reality applications.

Design of a 2.6 GHz GaN-HEMT Doherty Power Amplifier IC for Small-Cell Base Station Systems (Small-Cell 기지국 시스템을 위한 2.6 GHz GaN-HEMT Doherty 전력증폭기 집적회로 설계)

  • Lee, Hwiseob;Lim, Wonseob;Kang, Hyunuk;Lee, Wooseok;Lee, Hyoungjun;Yoon, Jeongsang;Lee, Dongwoo;Yang, Youngoo
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
    • /
    • v.27 no.2
    • /
    • pp.108-114
    • /
    • 2016
  • This paper presents a 2.6 GHz Doherty power amplifier IC to enhance the back-off efficiency. In order to apply to small-cell base stations, the Doherty power amplifier was fabricated using GaN-HEMT process for high power density. In addition, the implemented Doherty power amplifier was mounted on a QFN package. The implemented GaN-HEMT Doherty power amplifier was measured using LTE downlink signal with 10 MHz bandwidth and 6.5 dB PAPR for verification. A power gain of 15.8 dB, a drain efficiency of 43.0 %, and an ACLR of -30.0 dBc were obtained at an average output power level of 33.9 dBm.

A study on the new hybrid recurrent TDNN-HMM architecture for speech recognition (음성인식을 위한 새로운 혼성 recurrent TDNN-HMM 구조에 관한 연구)

  • Jang, Chun-Seo
    • The KIPS Transactions:PartB
    • /
    • v.8B no.6
    • /
    • pp.699-704
    • /
    • 2001
  • ABSTRACT In this paper, a new hybrid modular recurrent TDNN (time-delay neural network)-HMM (hidden Markov model) architecture for speech recognition has been studied. In TDNN, the recognition rate could be increased if the signal window is extended. To obtain this effect in the neural network, a high-level memory generated through a feedback within the first hidden layer of the neural network unit has been used. To increase the ability to deal with the temporal structure of phonemic features, the input layer of the network has been divided into multiple states in time sequence and has feature detector for each states. To expand the network from small recognition task to the full speech recognition system, modular construction method has been also used. Furthermore, the neural network and HMM are integrated by feeding output vectors from the neural network to HMM, and a new parameter smoothing method which can be applied to this hybrid system has been suggested.

  • PDF

Harmonics Control of Electric Propulsion System using Direct Torque Control (직접벡터제어방식을 사용하는 전기추진시스템의 고조파 제어)

  • Kim, Jong-Su;Oh, Sae-Gin
    • Journal of the Korea Institute of Information and Communication Engineering
    • /
    • v.13 no.12
    • /
    • pp.2618-2624
    • /
    • 2009
  • Harmonics (or distortion in wave form) has always existed in electrical power systems. It is harmless as long as its level is not substantial. However, with the recent rapid advancement of power electronics technology, so-called nonlinear loads, such as variable frequency drives for motor power/speed control, are increasingly finding their way to shipboard or offshore applications. In this paper a new approach to direct torque control (DTC) of induction motor drive is presented. In comparison with the conventional DTC methods the inverter switching frequency is constant and is dramatically increased, requiring neither any increase of the sampling frequency, nor any high frequency dither signal. The well-developed space vector modulation technique is applied to inverter control in the proposed DTC-based induction motor drive system, thereby dramatically reducing the current harmonics. As compared to the existing DTC approach with constant inverter switching frequency, the presented new approach does not invoke any concept of deadbeat control, thereby dramatically reducing the computations.

Convergence Property Analysis of Multiple Modulus Self-Recovering Equalization According to Error Dynamics Boosting (다중 모듈러스 자기복원 등화의 오차 역동성 증강에 따른 수렴 특성 분석)

  • Oh, Kil Nam
    • Journal of the Korea Academia-Industrial cooperation Society
    • /
    • v.17 no.1
    • /
    • pp.15-20
    • /
    • 2016
  • The existing multiple modulus-based self-recovering equalization type has not been applied to initial equalization. Instead, it was used for steady-state performance improvement. In this paper, for the self-recovering equalization type that considers the multiple modulus as a desired response, the initial convergence performance was improved by extending the dynamics of the errors using error boosting and their characteristics were analyzed. Error boosting in the proposed method was carried out in proportion to a symbol decision for the equalizer output. Furthermore, having the initial convergence capability by extending the dynamics of errors, it showed excellent performance in the initial convergence rate and steady-state error level. In particular, the proposed method can be applied to the entire process of equalization through a single algorithm; the existing methods of switching over or the selection of other operation modes, such as concurrent operating with other algorithms, are not necessary. The usefulness of the proposed method was verified by simulations performed under the channel conditions with multipath propagation and additional noise, and for performance analysis of self-recovering equalization for high-order signal constellations.

Performance Evaluation of the M-algorithm for Decoding Convolutional Codes (M-알고리듬을 이용한 컨벌루셔널 부호의 복호 성능 평가)

  • 천진영;최규호;성원진
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.27 no.3A
    • /
    • pp.188-195
    • /
    • 2002
  • The M-algorithm for decoding convolutional codes can significantly reduce the complexity of the Viterbi algorithm by tracking a fixed number of survivor paths in each level of the decoding trellis. It is an easily-implementable algorithm suited for real-time processing of high-speed data. The algorithm, however, generates a sequence of catastrophic errors when the correct path is not included in the set of survivor paths. In this paper, the performance of the M-algorithm obtained from using various decoding complexity levels, frame lengths, and code constraint lengths is presented. The performance gain is quantified when the algorithm is used in conjunction with codes of increased constraint length. In particular, it is demonstrated the gain from the increased code free distance overcompensates the loss from the correct path being excluded from the survivors, when the frame length is short to moderate. Using 64 survivor paths, the signal-to-noise ratio gain obtained by increasing the constraint length from K=7 to K=9, 11, 15 is respectively 0.6, 0.75, and 08dB, when the frame of length L=100 has the frame error rate of 0.01%.