• Title/Summary/Keyword: Adaptive noise cancellation

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Adaptive Phase Noise Compensator in Wireless ICS Repeater (무선 간섭 제거 중계기에서 적응형 위상 잡음 보상기 연구)

  • Jeong, Hae-Seong;Baek, Gwang-Hoon;Ryu, Heung-Gyoon
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.22 no.4
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    • pp.481-488
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    • 2011
  • In this paper, we study a novel wireless interference cancellation system(ICS) repeater based on OFDM system with phase noise analysis and performance evaluation. Recently, there were many researches about wireless repeater. The previous study is just considering the feedback channel cancellation algorithm. But, in wireless repeater system, the phase noise effect can exist at up and down converter of wireless repeater. In wireless repeater system, if there are phase noise effects, the performance of system will get worse. So, the phase noise compensator is needed in wireless repeater. Therefore, we propose adaptive phase noise compensator based on OFDM system in order to effectively cancel phase noise. In order to cancel interference, we adapt adaptive phase noise compensator in wireless repeater system. The adaptive phase noise compensator compensates phase noise effect. Therefore, in this paper, we analyze phase noise of wireless repeater based on OFDM system. In this paper, when phase noise power is -15 dBc and phase noise cutoff frequency is 100 kHz at 4QAM, the BER performance of propose algorithm is better about 4 dB than with adaptive equalizer and without phase noise compensator at $10^{-4}$.

A study on adaptive noise cancellation for enhancement of digital speech articulation (디지털음성명료도 향상을 위한 적응형 잡음제거 기법에 관한 연구)

  • Kim, Soo-Yong;Jee, Suk-Kun
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.11 no.5
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    • pp.961-968
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    • 2007
  • Today, we can use radio communication device anywhere-anytime. Sometimes, we use the device in acoustic noise environment. The acoustic noise makes many problems in communication system. In acoustic noise environment, speaker cannot send clear information to receiver, because the received signal includes both speech signal and noise signal. A digital filter is useful to remove noise to get desired signal. One of methods is the adaptive digital filter using the adaptive noise canceller that automatically adjust filter parameters. This thesis addresses articulation algorithms against actual acoustic noises by means of two adaptive filtering methods. One is the adaptive noise canceller with two input channels and another is the spectral subtraction filter with one input channel. The experimental result from the proposed filter shows that the adaptive noise canceller is useful to reduce the non-stationary noises, while the spectral amplitude filter is effective for stationary noises.

ADAPTIVE NOISE CANCELLATION APPLIED IN HYDRODYNAMIC FIELD

  • Liu, Yuanheng;Ma, Ning;Li, Tongyu
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1994.06a
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    • pp.802-805
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    • 1994
  • There are strong ocean wave interference with big amplitude very low frenqencies are similar to the ship's hydrodynamic signals. To detect ship's hydrodynamic field will be faced various natural hydrodynamic interferences which are radom and the prior knowlege of which are not know. This paper proposes to use the adaptive noise cancelling principle and used the model of adaptive wave canceller to eliminate the ocean wave interfrence and detect the ship's hydrodynamic signals. Computer simulation results shown that signal to noise ratio can be raised from several to ten times. It shows the fact that this mathod can detect the ship's hydrodynamic signals from the strong ocean wave interferences while it is difficult for the old methods.

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An Echo Canceller Robust to Noise and Residual Echo

  • Kim, Hyun-Tae;Park, Jang-Sik
    • Journal of information and communication convergence engineering
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    • v.8 no.6
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    • pp.640-644
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    • 2010
  • When we talk with hands-free in a car or noisy lobby, the performance of the echo canceller degrade because background noise added to echo caused by the distance from mouth to microphone is relatively long. It gives a reason for necessity of noise-robust and high convergence speed adaptive algorithm. And if acoustic echo canceller operated not perfectly, residual signal going through the echo canceller to far-end speaker remains residual echo, which degrade quality of talk. To solve this problem, post-processing needed to remove residual echo ones more. In this paper, we propose a new acoustic echo canceller, which has noise robust and high convergence speed, linked with linear predictor as a post-processor. By computer simulation, it is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint.

Convergence Analysis of a Filtered-x Least Mean Fourth Active Noise Controller (Filtered-x 최소평균사승 능동 소음 제어기 수렴분석)

  • 이강승
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1998.06d
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    • pp.80-83
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    • 1998
  • In this paper, we propose a new filtered-x least mean fouth (LMF) algorithm where the error raised to the power of four is minimized and analyze its convergence behavior or a multiple sinusoidal acoustic noise and Gaussian measurement noise. Application of the filtered-x LMF adaptive filter to active noise cancellation (ANC) requires estimating of the transfer characteristic of the acoustic path between the ouput and error signal of the adaptive canceller. The results of the convergence analysis of the filtered-x LMF algorithm indicates that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct component . Phase estimation error and estimated again. In particular , the convergence is shown to be strongly affected by the accuracy of the phase response estimate. Also, we newly show that convergence behavior can differ depending on the relative sizes of the Gaussian measurement noise and convergence constant.

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Enhancement of Convergence Speed of Adaptive Algorithm using Wavelet Packet Transform (웨이브렛 패킷 변환을 이용한 적응알고리듬의 수렴속도 향상)

  • 박서용;김대성
    • The Journal of Information Technology
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    • v.2 no.2
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    • pp.127-138
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    • 1999
  • The wavelet transform is widely used in signal processing application. In this paper, a wavelet domain adaptive algorithm(WPTNLMS) is derived and its performances are evaluated in non-stationary environment. Where the input signals are decomposed by the wavelet packet transform for the multi-resolution adaptive processing. And the NLMS is used as an adaptive algorithm in wavelet domain. The proposed technique is applied to noise cancellation of the Doppler signal which is added with white Gaussian noise.

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Sidelobe Cancellation Using Difference Channels for Monopulse Processing (모노펄스 처리용 차 채널을 이용한 부엽 잡음재머 제거)

  • Kim, Tae-Hyung;Choi, Dae-Young
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.26 no.5
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    • pp.514-520
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    • 2015
  • Sidelobe canceller(SLC) requires main beam pattern(SUM beam) and auxiliary beam patterns for rejection of sidelobe noise jammer. For best performance of sidelobe noise jamming cancellation of adaptive SLC, gain dominant region of each auxiliary beam pattern shall not be overlapped one another in elevation/azimuth regions of sidelobe of main beam, and beam patterns of auxiliary channels should have low gains in regions of mainlobe of main beam. In the monopulse radar, the difference beam patterns for monopulse processing have these properties. This paper proposes the method using data from the difference channel for monopulse processing as data from auxiliary channel for sidelobe cancellation. For the proposed SLC, the results of simulation and performance analysis was presented. If the proposed method is used in the monopulse radar, SLC can be constructed by using basic SUM and difference channels without extra channel composition.

Convergence of the Filtered-x LMS Algorithm for Canceling Multiple Sinusoidal Acoustic Noise (복수정현파 소음제거를 위한 Filtered-x LMS 알고리듬의 수렴 특성에 관한 연구)

  • Lee, Kang-Seung;Lee, jae-Chon;Youn, Dae-Hee;Kang, Young-Suk
    • The Journal of the Acoustical Society of Korea
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    • v.14 no.2
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    • pp.40-49
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    • 1995
  • Application of the filtered-x LMS adaptive filter to active noise cancellation requires to estimate the transfer charactersitics between the output and the error signal of the adaptive canceler. In this paper, we derive the filtered-x adaptive noise cancellation algorithm and analyze its convergence behavior when the acoustic noise consists of multiple sinusoids. The results of the convergence analysis of the filtered-x LMS algorithm indicate that the effects of the parameter estimation inaccuracy on the convergence behavior of the algorithm are characterized by two distinct components : Phase estimation error and estimated gain. In particular, the convergence is shown to strongly affected by the accuracy of the phase response estimate. Simulation results are presented to support the theoretical convergence analysis.

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Active noise control of a second-order Volterra system with an acoustic feedback path (음향 피드백 경로를 가진 2차 볼테라 시스템의 능동소음제어)

  • Lee, Jung-Jae;Kim, Kyoung-Jae;Seo, Jae-Bum;Nam, Sang-Won
    • Proceedings of the KIEE Conference
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    • 2008.04a
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    • pp.238-239
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    • 2008
  • In this paper, active noise control (ANC) of a Volterra system with a nonlinear secondary path is proposed in the presence of a linear acoustic feedback, whereby the conventional ANC of a linear system with online acoustic feedback-path modeling is further extended to ANC of a Volterra system with a linear acoustic feedback path. In particular, the proposed ANC system consists of two adaptive Volterra filters (for nonlinear noise control and nonlinear adaptive noise cancellation) and one feedback-path modeling filter. Simulation results show that the proposed approach yields more effective reduction of disturbances arising from the acoustic feedback, in addition to high nonlinear ANC performance.

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Hearing aid application of feedback cancellation algorithm in frequency domain (주파수 대역에서의 피드백 제거 알고리즘의 보청기 응용)

  • Jarng, Soon-Suck
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.4
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    • pp.272-279
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    • 2016
  • In this paper, the realization of a hearing aid adaptively cancelling feedback noise was considered. Conventional least mean square method in time domain was transformed into frequency domain in order to minimize computational burden. The adaptive filter algorithm was evaluated by Matlab (Matrix laboratory), and it was confirmed by CSR 8675 Bluetooth DSP IC (Digital Signal Processor Integrated Circuit) chip firmware realization. Some remote control features by a smart phone was added to the smart hearing aid for user interface easiness.