• Title/Summary/Keyword: 잡음 예측

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Performance Enhancement Technique of Visible Communication Systems based on Deep-Learning (딥러닝 기반 가시광 통신 시스템의 성능 향상 기법)

  • Seo, Sung-Il
    • The Journal of the Institute of Internet, Broadcasting and Communication
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    • v.21 no.4
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    • pp.51-55
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    • 2021
  • In this paper, we propose the deep learning based interference cancellation scheme algorithm for visible light communication (VLC) systems in smart building. The proposed scheme estimates the channel noise information by applying a deep learning model. Then, the estimated channel noise is updated in database. In the modulator, the channel noise which reduces the VLC performance is effectively removed through interference cancellation technique. The performance is evaluated in terms of bit error rate (BER). From the simulation results, it is confirmed that the proposed scheme has better BER performance. Consequently, the proposed interference cancellation with deep learning improves the signal quality of VLC systems by effectively removing the channel noise. The results of the paper can be applied to VLC for smart building and general communication systems.

A CELP Speech Coder Using Secondary Long Term Prediction with Multi-Band Pass Filtered Multi-Pulses (다중 펄스와 다중 대역 이차 장구간 예측을 이용한 CELP 음성 부호화기)

  • 서정태;최용수;강홍구;윤대희
    • The Journal of the Acoustical Society of Korea
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    • v.17 no.1
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    • pp.9-16
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    • 1998
  • 본 논문에서는 낮은 비트율 CELP 음성 부호화기의 장구간 예측기의 성능 향상 방 법을 제안한다. 비트율을 낮추기 위해서는 분석 구간의 길이가 길어져야하며 이에 따라 장 구간 예측기의 성능이 저하되어 장구간 예측 후에도 준 주기성 성분이 상당량 존재하므로 백색 잡음으로 구성된 통계 코드북만으로는 이를 모델링하기 어려워진다. 제안 방법에서는 다중 대역 필터와 다중 펄스열을 이용하여 한 번 더 필터링(이차 장구간 예측)함으로써 장 구간 예측 후의 신호가 통계 코드북에 적합한 백색 잡음 형태로 되도록 모델링한다. 제안된 방법의 성능을 평가하기 위해 4.8kbps 비트율로 양자화한 후, 기존에 제안된 같은 전송률의 MBCELP와 DoD-CELP와 비교하였다. 실험 결과 제안된 방법이 기존 부호화기들에 비해 주/객관적인 음질에서 우수한 성능을 보여준다.

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ELIMINATION OF BIAS IN THE IIR LMS ALGORITHM (IIR LMS 알고리즘에서의 바이어스 제거)

  • Nam, Seung-Hyon;Kim, Yong-Hoh
    • The Journal of Natural Sciences
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    • v.8 no.1
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    • pp.5-15
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    • 1995
  • The equation error formulation in the adaptive IIR filtering provides convergence to a global minimum regardless a local minimum with a large stability margin. However, the equation error formulation suffers from the bias in the coefficient estimates. In this paper, a new algorithm, which does not require a prespecification of the noise variance, is proposed for the equation error formulation. This algorithm is based on the equation error smoothing and provides an unbiased parameter estimate in the presence of white noise. Through simulations, it is demonstrated that the algorithm eliminates the bias in the parameter estimate while retaining good properties of the equation error formulation such as fast convergence speed and the large stability margin.

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A Study on the Adaptive Technique for Artifact Cancelling in Electroencephalogram Analysis System (뇌파 분석 시스템에서의 Artifact 제거를 위한 적응 기법에 관한 연구)

  • 유선국;김기만;남기현
    • Journal of Biomedical Engineering Research
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    • v.18 no.4
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    • pp.389-396
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    • 1997
  • Several types of electrical artifact seen on electroencephalogram( EEG) records are described. Those are the EOG and the PVC roller pump noise, and so on. An adaptive digital filtering of the electroencephalogram( EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, whore artifacts may not be acceptable in some cafes of automatic EEG processing. Thus we studied the method for cancelling these artifacts. This proposed method does not use the reference channel, and is realized by connecting the linear predictor and the fixed FIR filter for the EOG artifact, and by cascading the linear predictor and the noise canceller for the pump artifact. The simulation results illustrate the performances of the proposed method in terms of the capability of interferences suppression. In the results we obtained about 20 dB noise reduction.

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Noisy Time Varying Vibration Signal Analysis using Adaptive Predictor-Binary Tree Structured Filter Bank System (적응예측기-이진트리구조 필터뱅크 시스템을 이용한 잡음이 부가된 시변 진동신호 분석)

  • Bae, Hyeon-Deok
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.10 no.1
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    • pp.77-84
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    • 2017
  • Generally, a time-varying vibration signal is generated in a rotating machine system, and when there is a failure in the rotating machine, the signal contains noise. In this paper, we propose a system consisting of an adaptive predictor and a binary tree filter bank for analyzing time - varying vibration signals with noise. And the vibration signal analyzed results in this system is used for fault diagnosis of the rotating machine. The adaptive predictor of the proposed system predicts the periodic signal components, and the filter bank system decomposes the difference signal between the input signal and the predicted periodic signal into subband. Since each subband signal includes a noise signal component due to a failure, it is possible to diagnose the failure of the using rotary machine. The validity of the proposed vibration signal analysis method is shown in the simulations, where the periodic components cancelled vibrating signals are decomposed to 32 subband, and the signal characteristics related faults are analyzed.

Speech Recognition with Image Information (영상정보 보완에 의한 음성인식)

  • 이천우;이상원;양근모;박인정
    • Proceedings of the IEEK Conference
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    • 1999.06a
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    • pp.511-515
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    • 1999
  • The main factor decreasing speech recognition rate is the surrounding noise. To lower the noise effect, we generally used the filter bank at preprocessing stage. But, in this paper, we tried to recognize the 10 numeral numbers using 2-D LPC to extract image feature. At first, we obtained the result of speech-only recognition using 13th-order LPC coefficients and then, for distorted speech recognition results of ‘0’, ‘4’, ‘5’, ‘6’ and 9’, we added image parameters such as 12th-order 2-D LPC coefficients. At each frame, we extracted the 2-D LPC coefficients, and simulated recognizer with two parameters such as speech and image. Finally, for the numbers, such as ‘4’and ‘9’, the better results were obtained.

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Analysis of Measurement Accuracy Based on Confidences for Narrow-Band Underwater Acoustic Measurement (협대역 수중음향측정을 위한 신뢰도 기반의 측정정확도 분석)

  • 도경철;최재용;이용곤
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.4
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    • pp.16-22
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    • 2000
  • In order to predict the performance and the usefulness of the narrow-band underwater acoustic measurement system at design stage, whose error variance is not clearly described, in this study a boundary equation to estimate the measurement accuracy is proposed based on the confidency as SNR variation. The boundary is presented as a function of SNR and the number of samples. In this paper, the measurement performance for narrow-band signal is simulated by the proposed boundary equation and the results are reviewed in the biased noise condition and separately in the background noise rejected condition.

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Development of Interferences Reduction Algorithm for Ambulatory Blood Pressure Measurement (휴대용 혈압 측정을 위한 잡음 제거 알고리즘의 개발)

  • Choi, Hyun-Seok;Park, Ho-Dong;Lee, Kyoung-Joung
    • Proceedings of the KIEE Conference
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    • 2008.04a
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    • pp.131-132
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    • 2008
  • 오실로메트릭 방법으로 휴대용 혈압 측정 시 빈번하게 발생하는 잡음에 의한 오실레이션 신호의 왜곡을 줄이기 위해 새로운 잡음 제거 알고리즘을 제안하였다. 제안된 잡음 제거 알고리즘은 선형 예측기 구조 기반의 적응 필터를 이용한다. 제안된 잡음 제거알고리즘의 성능을 평가하기 위해 왜곡된 오실레이션 신호에 선형보간법을 사용하는 기존의 방법과 적응 필터를 사용하는 제안된 방법을 적용하여 잡음 제거 성능을 비교하였다. 제안된 방법은 잡음이 오실레이션과 중첩되어 나타난 경우에 기존의 방법과 달리 잡음에 강인한 특징을 보여주었다.

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Distance Measures Based Upon Adaptive Filtering For Robust Speech Recognition In Noise (잡음 환경하에서 음성 인식을 위한 적응필터링 거리 척도에 관한 연구)

  • 정원국;은종관
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.1E
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    • pp.15-22
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    • 1992
  • 잡음이 있는 환경하에서는 음성 인식의 성능이 현저하게 떨어지게 된다. 본 논문에서는 이렇나 잡음의 영향에 강한 거리척도를 제안하고자 한다. 우리는 잡음이 더해진 음성신호의 특징벡터를 깨끗한 음성신호의 특징벡터가 FIR 시스템을 거쳐 변형된 것이라고 가정한다. 여기서 FIR 시스템은 잡음의 영 향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템 계수잡음의 영향을 모델링한 것이라고 할 수 있다. 미지의 FIR 시스템계수들은 RLS 적응 알고리즘을 이용하여 구한다. 제안된 거리척도는 적응 여파 기의 예측 오차에 관한 식으로 표시되어진다. 여러 가지 적응 여파기의 구조중 단일 채널 일차 FIR 구 조가 가장 좋은 음성 인식 성능을 보이며, 이 경우 효과적인 거리척도 알고리즘을 구할 수 있다. 여러 가지 신호대 잡음비에 관하여 화자독립 격리단어 인식 실험을 DTW 알고리즘을 이용하여 수행하여 본 결과 제안된 거리척도가 거의 모든 신호대 잡음비에 대하여 우수한 성능을 보였다.

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Performance of the Recursive Systematic Convolutional Code with Turbo-Equalization Method for PMR Channel (수직자기기록 채널에서 터보등화기 구조를 이용한 순환 구조적 길쌈 부호의 성능)

  • Park, Dong-Hyuk;Lee, Jae-Jin
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.34 no.1C
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    • pp.15-20
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    • 2009
  • For perpendicular magnetic recording (PMR) channels, noise-predictive maximum likelihood (NPML) detection method has been used. But, it is hard to expect improving the performance when the bit density is increased. Hence, we exploit the coding methods which has good performance. In this paper, we show the performance of the recursive systematic convolutional (RSC) codes with turbo-equalization method with different channel bit densities. The noise model is 80% jitter noise and 20% AWGN.