• Title/Summary/Keyword: Channel Noise Time

Search Result 425, Processing Time 0.027 seconds

The Comparison of Simple Reaction Time between Young and Old Generation (청년층과 노인층의 단순반응속도 비교에 관한 연구)

  • Kwon Kyu-Sik;Choi Chul
    • Journal of Korean Society of Industrial and Systems Engineering
    • /
    • v.27 no.4
    • /
    • pp.133-140
    • /
    • 2004
  • This study deals with human reaction speed according to human physical conditions (body size) such as head width, thickness, breast width, arm extent, and age. Especially, the results of this study are compared between young and old generation. According to this study, the thickness and extent factor do not have any correlation with human reaction speed, but width factors(head width, breast width, etc) have some correlation with human reaction speed. The result of this study can be used to find fitter person for a special job such as emergency condition job, sports man (because you can find a person having a good talent for it without test). Also, the purpose of this study is to find CNT (Channel Noise Time). The word of CNT is to explain the relation between Channel Noise and working speed. (Channel Noise is a kind of noise happening between the human information transmission channel.)

Implementation of the single channel adaptive noise canceller using TMS320C30 (TMS320C30을 이용한 단일채널 적응잡음제거기 구현)

  • Jung, Sung-Yun;Woo, Se-Jeong;Son, Chang-Hee;Bae, Keun-Sung
    • Speech Sciences
    • /
    • v.8 no.2
    • /
    • pp.73-81
    • /
    • 2001
  • In this paper, we focus on the real time implementation of the single channel adaptive noise canceller(ANC) by using TMS320C30 EVM board. The implemented single channel adaptive noise canceller is based on a reference paper [1] in which it is simulated by using the recursive average magnitude difference function(AMDF) to get a properly delayed input speech on a sample basis as a reference signal and normalized least mean square(NLMS) algorithm. To certify results of the real time implementation, we measured the processing time of the ANC and enhancement ratio according to various signalto-noise ratios(SNRs). Experimental results demonstrate that the processing time of the speech signal of 32ms length with delay estimation of every 10 samples is about 26.3 ms, and almost the same performance as given in [1] is obtained with the implemented system.

  • PDF

A Study on the Reaction Speed according to Human Physical Condition (인간의 신체특성에 따른 반응속도에 관한 연구)

  • Kwon, Kyu-Sik;Choi, Chul
    • Science of Emotion and Sensibility
    • /
    • v.6 no.2
    • /
    • pp.9-16
    • /
    • 2003
  • This study deals with human reaction speed according to human Physical conditions (head width, breast width, arm extent etc). According to this study, the thickness and extent factor do not affect to human reaction speed, but width factor (head width, breast width). The result of this study can be used to find fitter person for a special job such as emergency condition job, sports man (because you can find a person having a good talent for it before the test). Also, the result of this study can be applied to make CNT (Channel Noise Time) clear The word of CNT is to explain the relation between Channel Noise and Time. (Channel Noise is a kind of noise in the human information transmission channel.).

  • PDF

An analysis of the short-term variation of the power line as a communication channel (전력선 통신 채널의 단 구간 변화에 대한 분석)

  • Park, Chong-yeon;Choi, Won-Ho;Jung, Kwang-Hyun
    • Journal of Industrial Technology
    • /
    • v.27 no.B
    • /
    • pp.21-27
    • /
    • 2007
  • The power line channel has time-variant characteristics caused by various kind of electrical devices. This characteristics are synchronized with the main voltage by their own characteristics. The main factors of disturbance are the variation of the channel impedance and noises. In other papers, the synchronous noise modeling has been achieved. But the modeling is not satisfied simultaneously with the time domain and the frequency domain and there are not any discussion about short-term variations of the channel impedance which cause to the signal fading. Therefore, this paper researched to solve problems about the signal fading by analyzing the short-term variation of the channel impedance, and proposed the synchronous noise modeling which is satisfied simultaneously in the time domain and the frequency domain.

  • PDF

Performance Improvement of Channel Estimation based on Time-domain Threshold for OFDM Systems (시간영역 문턱값을 이용한 OFDM 시스템의 채널 추정 성능 향상)

  • Lee, You-Seok;Kim, Hyoung-Nam
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.33 no.9C
    • /
    • pp.720-724
    • /
    • 2008
  • Channel estimation in OFDM systems is usually carried out in frequency domain based on the least-squares (LS) method and the minimum mean-square error (MMSE) method with known pilot symbols. The LS estimator has a merit of low complexity but may suffer from the noise because it does not consider any noise effect in obtaining its solution. To enhance the noise immunity of the LS estimator, we consider estimation noise in time domain. Residual noise existing at the estimated channel coefficients in time domain could be reduced by reasonable selection of a threshold value. To achieve this, we propose a channel-estimation method based on a time-domain threshold which is a standard deviation of noise obtained by wavelet decomposition. Computer simulation shows that the estimation performance of the proposed method approaches to that of the known-channel case in terms of bit-error rates after the Viterbi decoder in overall SNRs.

Mixture Filtering Approaches to Blind Equalization Based on Estimation of Time-Varying and Multi-Path Channels

  • Lim, Jaechan
    • Journal of Communications and Networks
    • /
    • v.18 no.1
    • /
    • pp.8-18
    • /
    • 2016
  • In this paper, we propose a number of blind equalization approaches for time-varying andmulti-path channels. The approaches employ cost reference particle filter (CRPF) as the symbol estimator, and additionally employ either least mean squares algorithm, recursive least squares algorithm, or $H{\infty}$ filter (HF) as a channel estimator such that they are jointly employed for the strategy of "Rao-Blackwellization," or equally called "mixture filtering." The novel feature of the proposed approaches is that the blind equalization is performed based on direct channel estimation with unknown noise statistics of the received signals and channel state system while the channel is not directly estimated in the conventional method, and the noise information if known in similar Kalman mixture filtering approach. Simulation results show that the proposed approaches estimate the transmitted symbols and time-varying channel very effectively, and outperform the previously proposed approach which requires the noise information in its application.

Performance Analysis of OFDM/QPSK System in Frequency Selective Rayleigh Fading Channel with Impulsive Noise (임펄스 잡음과 주파수 선택성 레일리 페이딩이 공존하는 통신로에서 OFDH/QPSK 시스템의성능 분석)

  • 조성언;박기식;김언곤;오원근;조경룡
    • Journal of the Korea Institute of Information and Communication Engineering
    • /
    • v.4 no.3
    • /
    • pp.643-652
    • /
    • 2000
  • In this paper, we analyze the BER performance of OFDM/QPSK system in frequency selective Rayleigh fading channel with impulsive noise and improve its performance by adopting convolutional coding. When the channel delay time is shorter than the guard band, the OFDM/QPSK system shows a good BER performance while, when the channel delay time becomes longer than the guard band, its BER performance is abruptly degraded. Moreover, when the transmitted signal is contaminated by a strong impulsive noise in the channel, the BER performance falls to about $10^{-1}$. Also, without channel coding technique, the system doesn't meet even the voice service requirement while it meets the data service requirement with convolutional coding in frequency selective Rayleigh fading channel with impulsive noise.

  • PDF

An Adaptive Mobility Estimator for the Estimation of Time-Variant OFDM Channels

  • Kim, Dae-jin;Kim, Cheol-Min;Park, Sung-Woo
    • Journal of Broadcast Engineering
    • /
    • v.6 no.1
    • /
    • pp.72-81
    • /
    • 2001
  • An adaptive channel estimation technique for OFDM-based DTV receivers is proposed using a new mobility estimator. Sample mean techniques for channel estimation have displayed good performance in slow fading channels, because averaging reduces noise In channel estimation operation. This paper suggests an algorithm which selects the optimal number of symbols within which the sample mean of consecutive pilot data can be obtained. The designed mobility estimator determines the optimal number by comparing mobility variance and estimated noise valiance. The algorithm using the mobility estimator obtains an optimal channel function under time-invariant or time-variant multipath fading channels, thereby making the best BER performance.

  • PDF

Channel estimation scheme of terrestrial DTV transmission employing unique-word based SC-FDE (Unique-word 채용한 SC-FDE 기반 지상파 DTV 전송의 채널 추정 기법)

  • Shin, Dong-Chul;Kim, Jae-Kil;Ahn, Jae-Min
    • Journal of Broadcast Engineering
    • /
    • v.16 no.2
    • /
    • pp.207-215
    • /
    • 2011
  • A signal passed through multi-path channel suffers ISI(Inter-Symbol Interference) and severe distortions caused by channel delay spread and noise components at the SC-FDE(Single Carrier with Frequency Domain Equalizer) transmission. Conventional UW(Unique-Word) based SC-FDE iterative channel estimation improves channel estimation performance by smoothing estimated CIR(Channel Impulse Response) of the noise components outside the channel length at time domain and restoring the broken cyclic property through UW reconstruction. In this paper, we propose channel estimation scheme through noise suppression within channel length. To suppress the noise, we estimate noise standard deviation as estimated CIR of the noise components outside the channel length and make criteria of the noise standard deviation gain that doesn't affect the original signal samples. When estimated CIR samples within channel length are less than the criteria value using the noise standard deviation and gain, the noise components are removed. Simulation results show that the proposed channel estimation scheme brings good channel MSE(Mean Square Error) and good BER(Bit Error Rate) performance.

64 Channel Noise Masking Digital Hearing Aid Firmware Development (64채널 소음 차폐 디지털 보청기 펌웨어 개발)

  • Jarng, Soon-Suck
    • The Journal of the Acoustical Society of Korea
    • /
    • v.31 no.6
    • /
    • pp.367-372
    • /
    • 2012
  • This paper introduces noise masking algorithm for 64 channel digital hearing aid. 125 Hz spectral resolution is maintained for 64 channels from 125 Hz to 8000 Hz. The same spectral masking processing effects as the cochlea are considered and applied for the present hearing aid noise reduction processing algorithm. Theoretical algorithm has been ported into assembler language program software and been embedded into a DSP IC chip for the digital hearing aid. Some parts of noise masking software program code are explained, and the results of the real-time noise reduction are verified by electro-acoustic measurements.