• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2004년도 하계종합학술대회 논문집(1)
• /
• pp.43-46
• /
• 2004

In this paper, we present a carrier phase estimation algorithm for LDPC coded systems. LDPC coded system can not achieve the ideal performance if phase offset is introduced by channel. However, the estimation of phase offset is very hard since the operating point of LDPC is very low SNR. To solve this problem, the algorithm using the tentative soft decision value and based on Maximum Likelihood (ML), was proposed in [2]. But this algorithm has problem which works only under constant phase offset. If phase offset is time variant, it has a severe degradation in performance. To solve this problem. we propose two types of estimators. symbol by symbol estimator: Unidirectional estimator (UDE) and hi-directional estimator (BDE), and sub-block estimator (SBE).

• 한국음향학회지
• /
• 제18권7호
• /
• pp.45-55
• /
• 1999

본 논문에서는 그레인 잡음을 제거하기 위해서 웨이브렛 변환(wavelet transform)에 근간을 둔 웨이브렛 적응 필터(WLMS adaptive filter : Wavelet domain Least Mean Square adaptive filter)를 사용하였다. 보통 그레인 잡음은 고온의 환경에서 금속의 결정구조가 변화함에 따라 발생된다. 웨이브렛 평면에서의 적응 필터링은 필터의 입력신호를 직교 변환하여 입력으로 이용함으로써 수렴 속도를 향상시킬 수 있는 장점을 가지고 있다. 적응 필터의 기준 입력 신호는 원시 입력 신호를 지연시킨 신호를 이용하였으며, 적응 필터의 출력은 다시 CA-CFAR(Cell Average - Constant False Alarm Rate) 임계 추정기(threshold estimator)를 거쳐 자동적으로 원하는 신호부분만 나타내도록 하였다. 우선 신호의 통계적 특성을 알기 위하여 run 테스트를 수행하여 기준 입력 신호가 비정상성(nonstationarity)을 나타냄을 보였고, 웨이브렛 적응필터가 시평면 적응필터보다 수렴속도면에서 우수함을 보였으며, 각 적응 필터의 출력신호에 대해서 신호대 잡음비를 통해 성능평가를 하였다. 시평면 적응 필터링 후에는 신호대 잡음비가 2-3㏈ 향상을 보였고, 반면 웨이브렛 적응 필터링후에는 신호대 잡음비가 4-6㏈ 향상을 보였다.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• 제13권2호
• /
• pp.962-977
• /
• 2019

The human voice is a convenient method of information transfer between different objects such as between men, men and machine, between machines. The development of information and communication technology, the voice has been able to transfer farther than before. The way to communicate, it is to convert the voice to another form, transmit it, and then reconvert it back to sound. In such a communication process, a vocoder is a method of converting and re-converting a voice and sound. The CELP (Code-Excited Linear Prediction) type vocoder, one of the voice codecs, is adapted as a standard codec since it provides high quality sound even though its transmission speed is relatively low. The EVRC (Enhanced Variable Rate CODEC) and QCELP (Qualcomm Code-Excited Linear Prediction), variable bit rate vocoders, are used for mobile phones in 3G environment. For the real-time implementation of a vocoder, the reduction of sound quality is a typical problem. To improve the sound quality, that is important to know the size and shape of noise. In the existing sound quality improvement method, the voice activated is detected or used, or statistical methods are used by the large mount of data. However, there is a disadvantage in that no noise can be detected, when there is a continuous signal or when a change in noise is large.This paper focused on finding a better way to decrease the reduction of sound quality in lower bit transmission environments. Based on simulation results, this study proposed a preprocessor application that estimates the SNR (Signal to Noise Ratio) using the spectral SNR estimation method. The SNR estimation method adopted the IMBE (Improved Multi-Band Excitation) instead of using the SNR, which is a continuous speech signal. Finally, this application improves the quality of the vocoder by enhancing sound quality adaptively.

• 전기학회논문지
• /
• 제67권1호
• /
• pp.124-130
• /
• 2018

The EMG amplitude estimator, which has been investigated as an indicator of muscle force, is of high relevance not only in biomechanical studies but also more and more in clinical applications. This paper presents a new approach to estimate surface EMG amplitude by using the mean spike and mean turn amplitude(MSA and MTA) variables. Surface EMG signals, a total of 198 signals, were recorded from biceps brachii muscle over the range of 20-80%MVC isometric contraction and performance of the MSA and MTA variables applied to amplitude estimation of the EMG signals were investigated. To examine the performance, a SNR(signal-to-noise ratio) was computed from each amplitude estimate. The results of the study indicate that MSA and MTA amplitude estimations with first order whitening filter and 300[ms]-350[ms] moving average window length are optimal and show better performance(mean SNR improvement of 6%-15%) than the most frequently used variables(ARV and RMS).

• Journal of information and communication convergence engineering
• /
• 제10권2호
• /
• pp.162-167
• /
• 2012

This paper proposes a speech processing system based on a model of the human auditory system and a noise reduction neural network with fast Fourier transform (FFT) amplitude and phase spectrums for noise reduction under background noise environments. The proposed system reduces noise signals by using the proposed neural network based on FFT amplitude spectrums and phase spectrums, then implements auditory processing frame by frame after detecting voiced and transitional sections for each frame. The results of the proposed system are compared with the results of a conventional spectral subtraction method and minimum mean-square error log-spectral amplitude estimator at different noise levels. The effectiveness of the proposed system is experimentally confirmed based on measuring the signal-to-noise ratio (SNR). In this experiment, the maximal improvement in the output SNR values with the proposed method is approximately 11.5 dB better for car noise, and 11.0 dB better for street noise, when compared with a conventional spectral subtraction method.

• Journal of Communications and Networks
• /
• 제16권6호
• /
• pp.583-591
• /
• 2014

In wireless communications, knowledge of the signal-to-noise ratio is required in diverse communication applications. In this paper, we derive the variance of the maximum likelihood estimator in the data-aided and non-data-aided schemes for determining the optimal shrinkage factor. The shrinkage factor is usually the constant that is multiplied by the unbiased estimate and it increases the bias slightly while considerably decreasing the variance so that the overall mean squared error decreases. The closed-form biased estimators for binary-phase-shift-keying and quadrature phase-shift-keying systems are then obtained. Simulation results show that the mean squared error of the proposed method is lower than that of the maximum likelihood method for low and moderate signal-to-noise ratio conditions.

• ETRI Journal
• /
• 제43권6호
• /
• pp.978-990
• /
• 2021

This article presents techniques for improved estimation of symbol timing offset (STO) and carrier frequency offset (CFO) for dual-polarization (DP) orthogonal frequency division multiplex (DP-OFDM) systems. Recently, quaternion multiple-input multiple-output OFDM has been proposed for high spectral efficiency communication systems, which can flexibly explore different types of diversities such as space, time, frequency, and polarization. This article focuses on synchronization techniques for DP-OFDM systems using a cyclic prefix, where the application of quaternion algebra leads to new improved estimators. Simulations performed for DP system methods show faster reduction of STO estimator variance with a double-slope line in the logvariance line versus signal-to-noise ratio (SNR) plot compared with singlepolarization (SP) counterparts, and simulations for CFO estimates show a 3-dB gain of DP over SP estimates for same SNR values defined, respectively, for quaternion-valued or complex-valued signals. Cramer-Rao bounds for STO and CFO are derived for the synchronization methods, correlating with the observed gains of DP over SP OFDM systems.

• 대한전자공학회:학술대회논문집
• /
• 대한전자공학회 2005년도 추계종합학술대회
• /
• pp.77-80
• /
• 2005

In this paper, we introduce the efficient carrier phase estimating algorithm collaborate with the channel decoder of turbo coded QPSK modulation for mobile DVB-RCS systems. At low SNR, the phase estimation using soft information of turbo decoder is able to improve power efficiency because of achieving the good synchronization. We investigate performance of external single estimator and internal multiple estimator in the PSP (Per Survivor Processing) manner over AWGN channel. For phase estimation, the LMS (Least Mean Square) scheme is considered. Three different APP-based methods are also proposed.

• 한국통신학회논문지
• /
• 제28권11C호
• /
• pp.1111-1117
• /
• 2003

이동 통신 시스템에서 이동체의 속도 추정은 수신신호의 최대 도플러 주파수를 찾는 과정이다. 하지만 실제적인 이동 통신 환경에서 이동체의 속도 추정은 부가된 잡음의 영향으로 많은 왜곡을 가지게 된다. 본 논문에서는 이동 통신 환경에서 최대 도플러 주파수 추정 시 부가 잡음의 영향을 억제하기 위한 결정 궤환 적응형 대역 제한(Decision Feedback Doppler Adaptive Band-Limit : DF-DABL) 알고리즘을 제안한다. 제안된 알고리즘에서는 잡음의 2차 통계 특성을 대역 제한 방법을 통하여 신호의 특성과 일치하게 함으로서 부가된 잡음의 영향을 효과적으로 제거한다. 특히 제안된 방법에서는 신호대잡음비 (SNR)와 같은 추가 채널 정보를 필요로 하지 않는다.

• Journal of information and communication convergence engineering
• /
• 제5권3호
• /
• pp.185-193
• /
• 2007

This paper evaluates the performance combining space-time transmit diversity (STTD) and an efficient channel estimator (ECE) for wideband code division multiple access (WCDMA) systems in various mobile channels. Using decision variable (DV), we also derive the analytic bit error rate (BER) and mean square error (MSE) for WCDMA applying ECE for STTD schemes. The simulation results show that the ECE performance is superior to the previous works in [1] as because we use additional pilot diversity which is so called secondary common control physical channel (S-CCPCH). The performance in case of the channel estimator using only one-channel or two-channel is worse than that of an ECE as about the maximum 4 dB at BER 1.0E-3 satisfying voice service over Rician fading channel. Our results show that, even with ICE, an ECE algorithm are effective in improving the output SNR and significantly reduce the error floor. In addition, the simulation results investigated in this paper also reveal that WCDMA combining an ECE and the STTD scheme could provide appreciable performance improvements in Rayleigh fading channel.