• Title/Summary/Keyword: Noise Cancelling

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Noise Cancellation using Cascaded Adaptive Filtering in EEG (직렬로 연결된 적응 필터링을 이용한 EEG내의 잡음 제거)

  • Kim, K.M.;Yoo, S.K.;Kim, N.H.
    • Proceedings of the KOSOMBE Conference
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    • v.1996 no.05
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    • pp.143-146
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    • 1996
  • An adaptive digital filtering of the electroencophalogram(EEG) is a successful way of suppressing mains interference, but it affects some of the frequency components of the signal, an artifact which not be acceptable in some cases of automatic EEG processing. The types of electrical artifact seen on EEG records is described. Those are the EOG and the PVC roller pump noise. And we study the method for cancelling these artifacts. The method does not need the reference channel, and are obtained by cascading the linear predictor and the noise canceller. The simulation results illustrate the performances of the presented method in terms of the capability of interferences suppression.

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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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A Study on the Fast QR RLS Algorithm for Applications to Adaptive Signal Processing (적응 신호 처리에의 응용을 위한 고속 QR RLS 알고리즘의 연구)

  • 정지영
    • Proceedings of the Acoustical Society of Korea Conference
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    • 1991.06a
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    • pp.38-41
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    • 1991
  • RLS algorithms are required for applications to adaptive line enhancers, adaptive equalizers for voiceband telephone and HF modems, and wide-badn digital spectrum mobile raio in which their convergence time and tracking speed are significant. The fast QR RLS algorithm satisfies above the requirements. Its computational complexity is linearly proportional to the tap number of a filter, N and its performance remains numerically stable. From the result of simumulation, the fast QR RLS algorithm represented Cioffi is better than gradient based algorithm in its initial performance when being applied to an adaptive line enhancer for cancelling noise.

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A Design of ANC-ALE Model Using the JP Lattie Filter (JP 격자필터를 이용한 ANC-ALE 모형 설계)

  • 정준철;심수보
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.16 no.12
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    • pp.1219-1228
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    • 1991
  • In the actual case, a model of noise canceller using adaptive filter has both a channel transfer function from noise source to main signal input and to noise canceller input. The previous models of noise canceller have been considered to be only one side channel transfer function. Therefore, it is proposed that a new model has two channel transfer functions and derives an optimal tranfer function of adaptive noise canceller. The adaptive filter is using the joint process lattice filter that has fast adaptive speed. The signal noise radio has been improved by a model of ANC-ALE and it is confirmed with computer simulation. Beside, a dc bias is very effective for noise cancelling, especially to the particular signal.

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Noise Cancelling in Multiple Noise Sources Using Correlation Elimination Filter (상관제거 필터를 이용한 다중 소음원의 소음제거)

  • Jee, Suk-Kun;Kim, Chun-Duck
    • The Journal of the Acoustical Society of Korea
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    • v.13 no.6
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    • pp.15-23
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    • 1994
  • This paper described adaptive ANC (Active Noise Control) systems for noise canacelling in multiple inputs and multiple outputs channel. Until this time, it was reported that the study of AMC for the multiple inputs and the multiple outputs had no correlation between input signals of adaptive filters. In fact, it is difficult independently to check the noise source signals and insufficient about this study. In this paper we proposed to use the adaptive filter for eliminating the correlation between input signals. The simulation results showed that the convergence rate improved about two times with this filter than that one without this filter at the rate $b_{12}=0.95$.

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A Study on "A Midsummer Night's Palace" Using VR Sound Engineering Technology

  • Seok, MooHyun;Kim, HyungGi
    • International Journal of Contents
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    • v.16 no.4
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    • pp.68-77
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    • 2020
  • VR (Virtual Reality) contents make the audience perceive virtual space as real through the virtual Z axis which creates a space that could not be created in 2D due to the space between the eyes of the audience. This visual change has led to the need for technological changes to sound and sound sources inserted into VR contents. However, studies to increase immersion in VR contents are still more focused on scientific and visual fields. This is because composing and producing VR sounds require professional views in two areas: sound-based engineering and computer-based interactive sound engineering. Sound-based engineering is difficult to reflect changes in user interaction or time and space by directing the sound effects, script sound, and background music according to the storyboard organized by the director. However, it has the advantage of producing the sound effects, script sound, and background music in one track and not having to go through the coding phase. Computer-based interactive sound engineering, on the other hand, is produced in different files, including the sound effects, script sound, and background music. It can increase immersion by reflecting user interaction or time and space, but it can also suffer from noise cancelling and sound collisions. Therefore in this study, the following methods were devised and utilized to produce sound for VR contents called "A Midsummer Night" so as to take advantage of each sound-making technology. First, the storyboard is analyzed according to the user's interaction. It is to analyze sound effects, script sound, and background music which is required according to user interaction. Second, the sounds are classified and analyzed as 'simultaneous sound' and 'individual sound'. Thirdly, work on interaction coding for sound effects, script sound, and background music that were produced from the simultaneous sound and individual time sound categories is done. Then, the contents are completed by applying the sound to the video. By going through the process, sound quality inhibitors such as noise cancelling can be removed while allowing sound production that fits to user interaction and time and space.

Secondary Path Estimation Algorithm Based on Residual Music Canceller for Noise Cancelling Headphone (노이즈 캔슬링 헤드폰에 적합한 잔여 음악 제거기 기반의 2차 경로 추정 알고리즘)

  • Ji, Youna;Lee, Keunsang;Park, Youngcheol
    • The Journal of the Acoustical Society of Korea
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    • v.34 no.5
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    • pp.377-384
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    • 2015
  • An active noise control (ANC) algorithm for noise canceling headphone is proposed. In this study, the feedback ANC operated with the filtered-x least mean square algorithm (FxLMS) algorithm is used to attenuate the undesired noise. Also an adaptive residual music canceller (RMC) is proposed for enhancing the accuracy of the reference signal of the feedback ANC. Simulation results show that a high quality of music sound can be consistently achieved in a time-varying secondary path situation.

A implementation on active silencer using hysteresis control (히스테리시스 제어기법을 이용한 능동 소음기의 구현)

  • 강정유;민휘식;황희융;장도현
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.34S no.7
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    • pp.112-122
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    • 1997
  • This paper presents active silencing by hysteresis control method and shows its audible noise reduction effect. A Pentium processor and sound blaster 16 card are used for its implementation. The Sound-Blaster 16 executes the A/D, D/A conversion and is used for operating source of lodspeaker for cancelling. As a result, between the frequency range of 100[Hz] and 200[Hz] there is an audible noise reduction effect from 9[dB] to 15[dB], and between the 300[Hz] and 500[Hz] range there is a reduction from 3[dB] to 5[dB].

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A study on the permanent levitation system for Maglev train (자기부상열차의 영구자석 부상계에 대한 연구)

  • Moon, Seok-Jun;Yun, Dong-Won;Cho, Hung-Je;Park, Sung-Whan;Kim, Byung-Hyun
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2007.11a
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    • pp.82-87
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    • 2007
  • A Halbach array is a special arrangement of permanent magnets which augments the magnetic field on one side of the device while cancelling the field to near zero on the other side. The application of this Halbach array magnet to the electrodynamic suspension for Maglev train has been recently studied in order to increase the levitation capability. This paper is focused on analytical method of the magnetic levitation system using Halbach array magnet. The suitability of the proposed method is verified with comparing to a finite element method. From this study, it is confirmed that the proposed method provides a reasonable solution with a little analysis time to the finite element method and the magnetic levitation system using Halbach array magnet is stable dynamically.

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Modified FxLMS Algorithm for Active Noise Control and Its Real-Time Implementation

  • Mu, Xiangbin;Ko, JinSeok;Rheem, JaeYeol
    • Journal of the Institute of Electronics and Information Engineers
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    • v.50 no.9
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    • pp.172-176
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    • 2013
  • This paper presents a modified filtered-x least mean square (FxLMS) algorithm to improve the stability of active noise control (ANC) system in realistic environment. A real-time ANC system employing modified FxLMS is designed and implemented on digital signal processor (DSP) board. The ANC system is evaluated for cancelling various tonal frequency noises in the range from 100 to 500 Hz and the performance is measured in terms of sound pressure level (SPL) attenuation. Experiment results show that a quiet zone with maximum 20 dB SPL attenuation can be generated around the location of error microphone.