• Title/Summary/Keyword: Signal Canceller

Search Result 185, Processing Time 0.025 seconds

Adaptive Filter for Noise Cancellation of ECG's (심전도 신호의 잡음 제거를 위한 적응 필터)

  • Lee, Jae-Joon;Song, Chul-Gyu;Lee, Je-Suk;Lee, Myoung-Ho
    • Proceedings of the KOSOMBE Conference
    • /
    • v.1992 no.05
    • /
    • pp.186-189
    • /
    • 1992
  • Adaptive fliter for noise cancellation of ECG is proposed. An adaptive noise canceller using the least mean squares algorithm is used to reduce unwanted noise. The adaptive noise canceller minimizes the mean-square error between a primary input, which is the noisy ECG, and a reference input, which is either noise that is correlated in some way with the noise in the primary input or a signal that is correlated only with ECG in the primary input.

  • PDF

Performance Analysis of a Cascaded Interference Canceller Employing Soft Detection for DS/CDMA System (DS/CDMA 시스템에서 소프트검출 방식을 이용한 직렬 간섭 제거기에 대한 성능분석)

  • Lee, Sang-Hoon;Kim, Nam
    • Proceedings of the IEEK Conference
    • /
    • 1999.06a
    • /
    • pp.849-852
    • /
    • 1999
  • In this paper, the performance of DS/CDMA receiver is analyzed, which employs a cascaded interference canceller using a soft detection. The channels considered is characterized by the multipath fading which is one of the factors for the performance degradation in mobile communication systems. The idea behind the cancellation is that the co-channel interference can be regenerated at the receiver and is subtracted from the received signal. The numerical results show the improvement of BER can be obtained by the proposed cancellation scheme.

  • PDF

A study on improvement of steady-state peformance and convergence rate in an adaptive noise canceller (적응잡음제거기의 정상상태 성능 및 수렴율 향상에 관한 연구)

  • 배종갑;김창기;박장식;손경식
    • Journal of the Korean Institute of Telematics and Electronics S
    • /
    • v.34S no.4
    • /
    • pp.42-49
    • /
    • 1997
  • A conventional adaptive noise canceller (ANC) using LMS algorithm suffers from the misadjustment of adaptive filter weights due to the gradient-estimate noise by input speech signal at steady state. In this paper, an ANC is proposed which uses the combination of VSLMS (variable step size LMS) and SA (sign algorithm) to improve steady state performance and convergence rate. SA algorithm is applied in speech region to prevent the weights from perturbing by output speech of ANC and VSLMS algorithm is applied to improve convergence rate and channel tracking ability in silence region and adaptive transient region. In compute rsimulation, the performance of the proposed VSLMS-SA combination algorithm is much better than LMS algorithm and the algorithm, recently proposed by greenberg, with adaptation step-size parameter determine dby sum method in convergence rate, channel tracking and steady state performance.

  • PDF

Timing Titter Analysis in the ISDN U-Interface (ISDN U-Interface에서 타이밍지터의 해석)

  • 김동관;이명수;강창언
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.13 no.5
    • /
    • pp.369-378
    • /
    • 1988
  • In this paper, the performance of the timing jitter which has great effects on the echo canceller that can be used for full-duplex digital transmission on two-wire subscriber loops is analyzed. The power spectrum of timing jitter is about 8.9dB lower in the AMI input format than in the Polar-NRZ-L input format. The performance of the echo canceller also has been shown improved by 4dB when the input signal is in the AMI format.

  • PDF

NBI Rejection Techniques using Improved Decision Feedback for DS/SS Systems (DS/SS 시스템을 위한 개선된 결정궤환 구조를 가지는 협대역 간섭신호 제거)

  • 유창현;시광규
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.21 no.10
    • /
    • pp.2679-2686
    • /
    • 1996
  • In this paper, we propsoed the two methods to improve the conventional decision feedback interference canceller in DS/SS communication systems. The data bit is obtained by correlating the PN sequence with the received signals to the present time k, and thus the errors in the reference signal can be reduced by newly deciding all the reference signals with the resultant data bit. Additionally the cancelled signals are computed with less weight for initial reference signals of low processing gain, and highly weighted as the processing gain goes up. the resulting interference canceller outperforms the existing ones. By simulation, we found the proposed algorithm has "2-3 dB" performance gain at BER 10$^{-3}$ compared to the conventional descision feedback algorithm.algorithm.

  • PDF

A Modified Robust Adaptive Beamformer for Microphone Arrays

  • Lee, Young-Ho;Choi, Su-Young;Park, Jans-Sik;Son, Kyung-Sik
    • Proceedings of the Korea Multimedia Society Conference
    • /
    • 2003.05b
    • /
    • pp.446-449
    • /
    • 2003
  • The conventional GSC is inappropriate in real situation when the target signal is present. The steering vector error cancels the target signal and the target signal misadjusts the weight of the adaptive filter. To prevent the target signal cancellation, the robust GSC using the constrained adaptive filters was already proposed. However, the adaptive weight misadjustment is not settled in robust GSC. This Paper proposes a revised robust sidelobe canceller with adaptive compensator. To compensate the influence of target signal, the adaptive compensator is used in cascade. In computer simulation, we show the performance improvement by comparing the robust GSC with the proposed GSC.

  • PDF

Noise Reduction of PPG Signal During Free Movements Using Adaptive SFLC(Scaled Fourier Linear Combiner) (적응 SFLC(Scaled Fourier Linear Combiner)를 이용한 활동 중의 PPG 신호의 잡음 감소)

  • Kim, Sung-Min;Cha, Eun-Jong;Kim, Deok-Won;Yoo, Jae-Ha;Kim, Dong-Yon;Kim, Soo-Chan
    • The Transactions of the Korean Institute of Electrical Engineers D
    • /
    • v.55 no.3
    • /
    • pp.138-141
    • /
    • 2006
  • Blood flow is one of vital signals related to human physiological information. Photoplethysmograph (PPG) has been used to measure indirectly heart rate, blood oxygen saturation ($SpO_2$), and so on. Because PPG signal is weak and sensitive to motion artifacts, it is very important to continuously obtain stable PPG signal during free movement. In this study, we applied the scaled Fourier linear combiner (SFLC) using both the adaptive filter and FLC to remove effectively the motion artifacts as well as background noise in the real time without additional signal correlated with motion from a accelerometer. The proposed method would be useful to reduce the movement and background noise which are not synchronized with heart rate.

A Noise Robust Adaptive Algorithm for Acoustic Echo Caneller

  • Lee, Young-Ho;Park, Jeong-Hoon;Park, Jang-Sik;Son, Kyong-Sik
    • Proceedings of the Korea Multimedia Society Conference
    • /
    • 2003.05b
    • /
    • pp.423-426
    • /
    • 2003
  • Adaptive algorithm used in Acoustic Echo Canceller (AEC) needs fast convergence algorithm when reference signal is colored speech signal. Set-Membership Affine Projection (SMAP) algorithm is derived from the constraint, which is the minimum value adaptive filter coefficient error. In this paper, we test the characteristic about noise of the SMAP algorithm and proposed modified version of SMAP algorithm fur using at AEC. As the projection order increase, the convergence characteristic of the SMAP algorithm is improved where no noise space. But if the noise uncorrelated with input signal exists, the AEC shows bad performance. In this paper, we propose normalized version of adaptive constants using estimated error signal for robust to noise and show the good performance through AEC simulation.

  • PDF

Implementation of Acoustic Echo Canceller with A Post-processor Using A Fixed-Point DSP (고정 소수점 DSP를 이용한 후처리기를 가지는 음향 반향제거기의 구현)

  • 이영호;박장식;박주성;손경식
    • Journal of Korea Multimedia Society
    • /
    • v.3 no.3
    • /
    • pp.263-271
    • /
    • 2000
  • In this paper, an acoustic echo canceller(AEC) is implemented by ADSP-2181. This AEC uses a noise robust adaptive algorithm and a postprocessing method which attenuates residual echo using cross-correlation between estimated error signal and microphone input signal. We propose new postprocessing method that uses two thresholds to prevent signal distortion after postprocessing and to improve the performance of AEC without extra computational burden. Through experiments using a 16 bit fixed-point DSP board (ADSP-2181 EZ-KIT Lite board), it is shown that the noise robust adaptive algorithm performs well in the double-talk situations and the convergence speed is comparable to NLMS. Using the postprocessor, ERLE is improved about 20 dB. As a result, the AEC with a postprocessor shows better performance than conventional ones.

  • PDF

Double Talk Detection using the Fuzzy Inference (퍼지 추론을 이용한 동시통화 검출)

  • 류근택;배현덕
    • Journal of Broadcast Engineering
    • /
    • v.5 no.1
    • /
    • pp.123-129
    • /
    • 2000
  • This paper addresses a new double detection algorithm which is based on the fuzzy control in the adaptive echo canceller of communication system. In this method, the two input of the fuzzy inference for detecting double talk condition are used. The one is the cross-correlation coefficient between the error signal and the primary signal which is the summed signal of the real echo signal and the near-end signal. The other is the cross-correlation coefficient between the estimation error signal and the primary signal. The fuzzy controller made a fuzzification for two inputs by the membership functions of trapezoid and them became the composition using inference rules. The composed result is defuzzificated by the center gravity method. The output is compared with two threshold values to detect double talk and echo path variation effectively. It is confirmed by computer simulation that this fuzzy double talk detector is able to track echo path variation accurately.

  • PDF