• Title/Summary/Keyword: Acoustic filter(음향 필터)

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A Study on the Transaural Filter Implementation for 5.1 Channel Speaker System (5.1채널 스피커 시스템에서 트랜스오럴 필터 구현에 관한 연구)

  • 최갑근;방승범;김순협;정완섭
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.3
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    • pp.245-255
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    • 2002
  • This thesis deals a method to deliver more realistic sound by cancelling the cross-talk which is inherent to the 5.1 channel speaker system. The acoustical model for cross-talk cancellation is the free field model. This model minimizes distortion of sound. I used the bark scale sound quality compensation which based on psycho-acoustic. For the surround channels, band-limited sound quality compensation is performed in the frequency domain. I also performed the sound quality assessment test on the traditional 2 channel stereo and 5.1 channel system. This test is performed in the test chamber which satisfies the ITU-R specifications. I uses the IACC (Inter-Aural Cross-Correlation) to determine the preferences of the amateur and the golden ear experts to asses the trans-aural filter. According to the result from the proposed method, I got more the 38 dB separation rates with the Dolby standard speaker array. The results on the diffusion by the subjective test with the experts shows 0.4 point increased then before.

A Study on the Subband Acoustic Echo Canceller Using Weighted Overlap-Add SSB and QMF Filter Banks (중첩가산방식의 SSB 필터뱅크와 QMF 필터뱅크를 이용한 서브밴드 음향 반향 신호 제거기에 관한 연구)

  • 차경환;심동연;김천덕
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.36S no.4
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    • pp.93-100
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    • 1999
  • 확성회의 시스템에서 응용되는 반향신호 제거기는 긴 잔향시간을 갖는 실내 공간의 환경변화에 따라 필터 계수의 갱신에 많은 시간이 요구되어 실시간 처리에 문제점으로 지적되고 있다. 본 논문에서는 연산량 저감을 통한 실시간 처리를 위하여 중첩가산방식의 SSB(Single Side Band) 필터뱅크를 사용한 서브밴드 적응 신호처리법을 제안한다. 이 방법은 입력과 출력의 스펙트럼을 몇 개의 주파수 밴드로 분할하여, 각 밴드를 ES-NLMS(Exponential Step-Normalized Least Mean Square) 알고리즘을 이용하여 적응 처리하는 것이다. 시뮬레이션 결과 중첩가산방식의 SSB 필터뱅크가 풀밴드 보다 ERLE(Echo Return Loss Enhancement)가 1∼2㏈ 정도 작을 때 연산량이 풀밴드 보다 약95%, QMF(Quadrature Mirror Filter)필터뱅크보다 약50% 정도 감소하여 우수한 것으로 나타났다.

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Matched filter Using Acoustoelectric Memory Convolver (Acoustoelectric 기억 콘벌버를 이용한 정함필터)

  • 최영호;정영지;황금찬
    • The Journal of the Acoustical Society of Korea
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    • v.3 no.2
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    • pp.13-22
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    • 1984
  • A surface acoustic wave signal processing device using the silicon surface state is presented and shown capable of storing a reference signal and later correlating another signal with the stored reference. The device memory consists of the storage of the spatial 2k pattern of an acoustic wave as stored charges in the surface state of silicon surface. Results of experiments are presented which characterize the operation of device. Simpliied models for charging process and nonlinear acoustoelectric interactions based on consideration of single surface state at the surface of silicon The validity of simplified model has been qualitatibely confirmed with experimental results and the application of this device to aprogrammable matched filter of communication is considered.

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FBAR devices for RF bandpass filter applications (RF 대역통과필터 응용을 위한 FBAR 소자)

  • Giwan Yoon;Park, Sungchang
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2001.10a
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    • pp.621-625
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    • 2001
  • In this article, piezoelectric films and their application for film bulk acoustic resonator (FBAR) devices are presented. The FBAR is composed of piezoelectric film sandwiched between top and bottom electrodes and an acoustic reflector of SiO$_2$/W slatted multilayers. Various FBAR devices were fabricated and evaluated through simulation and measurement. The insertion loss, return loss and Q-factor were observed to be reasonably high and good. The FBAR technology seems very promising particularly for RF band filter application.

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An Algorithm on Optimum Weighting Design in Beamforming for Acoustic Measurement (음향측정을 위한 빔형성에서의 최적 가중상수 설계 기법)

  • Dho, Kyeong-Cheol;Son, Kweon;Lee, Yong-Gon;Son, Kyung-Sik
    • The Journal of the Acoustical Society of Korea
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    • v.18 no.8
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    • pp.61-67
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    • 1999
  • This paper proposes a new beamforming algorithm for acoustic measurement by using the nested linear array. In this algorithm, the weighting is optimized by minimizing the LMS error with the initial value obtained by FIR filter design algorithm. The optimization process is applied to each sub-band, which is divided from the octave band, to produce the uniform directivity index. For the optimization pseudo inverse matrix is used for the transfer matrix. As the simulation results, it is found that the proposed algorithm can get the desired beam pattern and unform directivity index so as to be used efficiently for the acoustic measurement by using a nested linear array.

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Applying an Auxiliary Filter in the Adaptive Echo Canceller for Performance Improvement of Double-Talk Detection (음향반향제거기에서 동시통화 검출 성능 개선을 위한 보조필터 적용)

  • Kim Siho;Bae Keunsung
    • Journal of the Institute of Electronics Engineers of Korea SP
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    • v.42 no.1
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    • pp.65-70
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    • 2005
  • This paper deals with the problem of double-talk (DT) detection in anacoustic echo canceller (AEC). In the DT detection algorithm with correlation coefficient, detection errors occasionally occur because it is hard to set the threshold to distinguish DT from echo path change (EPC). Adaptive filter falls into the situation that it stops updating its filter coefficients when EPC is erroneously considered as DT at the starting-point of EPC. In addition, in case of echo path changing during the DT period, the end-point detection of DT period fails so that the AEC cannot update its filter coefficients for a while even after the DT period ends. To solve these problems, in this paper, we propose a novel AEC that employs an auxiliary filter. We use the idea that though the error signal cannot be estimated using reference signal in case or DT situation but it can be in case or EPC situation. The experimental result verifies that the proposed method could solve the problems caused by DT detection error or echo path change during the DT period.

Design of Acoustic Source Array Using the Concept of Holography Based on the Inverse Boundary Element Method (역 경계요소법에 기초한 음향 홀로그래피 개념에 따른 음원 어레이 설계)

  • Cho, Wan-Ho;Ih, Jeong-Guon
    • The Journal of the Acoustical Society of Korea
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    • v.28 no.3
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    • pp.260-267
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    • 2009
  • It is very difficult to form a desired complex sound field at a designated region precisely as an application of acoustic arrays, which is one of important objects of array systems. To solve the problem, a filter design method was suggested, which employed the concept of an inverse method using the acoustical holography based on the boundary element method. In the acoustical holography used for the source identification, the measured field data are employed to reconstruct the vibro-acoustic parameters on the source surface. In the analogous problem of source array design, the desired field data at some specific points in the sound field was set as constraints and the volume velocity at the surface points of the source plane became the source signal to satisfy the desired sound field. In the filter design, the constraints for the desired sound field are set, first. The array source and given space are modelled by the boundary elements. Then, the desired source parameters are inversely calculated in a way similar to the holographic source identification method. As a test example, a target field comprised of a quiet region and a plane wave propagation region was simultaneously realized by using the array with 16 loudspeakers.

Stereo Echo Canceller Using Non-linear Pre-processing Filter (비선형 전처리 필터를 이용한 스테레오 음향반향 제거기)

  • 정일규;김현태;박장식;손경식
    • Proceedings of the IEEK Conference
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    • 2000.09a
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    • pp.193-196
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    • 2000
  • An stereophonic acoustic echo canceller cannot exactly estimate the echo path in the receiving room, because of the cross-correlation between stereo signals. In this paper, the new preprocessing filter is proposed to reduce the cross-correlation between the signals without influence on stereophonic sound. Two channel signals are linearly decorrelated by using orthogonality principles and the attenuated absolute values of the decorrelated signals are added to each channel input signals. Assuming that the power of each channel signal is larger than that of the cross-correlation, computational burden is reduced.

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Design of Small Acoustic Filter for ITE Type Noise Protector (ITE 타입의 소음 차폐기용 소형 음향 필터의 설계)

  • Lee, Yun-Jung;Kim, Pil-Un;Chang, Yong-Min;Lee, Sang-Heun;Cho, Jin-Ho;Kim, Myoung-Nam
    • Journal of Sensor Science and Technology
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    • v.13 no.6
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    • pp.479-487
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    • 2004
  • The prevention of noise induced hearing loss is very important, because there is no treatment for it. There are some kinds of devices for hearing protection, and those are effective in preventing a noise induced hearing loss. However, people often resist the use of hearing protection devices because it is difficult to have a conversation. Therefore, a hearing protection device is must effective not only in hearing protection but also in preserving communication ability. In this paper, we proposed a small acoustic filter for ITE type noise protector to solving the problem about the difficulties of conversation. That is applied a principle of acoustic filters that have been used for a muffler of automobiles, guns, and etc. To find out the sound transmission characteristic at the eardrum, we regarded an acoustic filter and external ear canal as a coupled system. So, we simulated the coupled system with OrCad, and experimented with a designed acoustic filter and a 2 cc coupler which has the same transmission characteristic as the external ear canal has. We confirmed that it is possible to adjustment acoustic transmission characteristics through simulation of electrical model for acoustic filter and external ear and experiments using designed small acoustic filters.

Improving a Sound Localization Using 1/3-octave Band Pass Filter (1/3-옥타브 대역통과필터를 이용한 음상정위기법 성능 향상)

  • Hwang, Shin;Yang, Jin-Woo;Cheung, Wan-Sup;Kim, Soon-Hyob
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.3
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    • pp.98-103
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    • 2001
  • The binaural auditory system of human has the capability of differentiating the direction and distance of sound sources. This feature is well characterised in terms of the inter-aural intensity difference (IID), the inter-aural time difference (ITD) and/or the spectral shape difference (SSD) arising from the acoustic transfer of a sound source to the outer ears. This paper proposes an effective way of extracting the three sound perception factors (IID, ITD, SSD) from the head-related transfer functions (HRTF's) that depends on the direction and distance of the acoustic source from the listener. It includes the estimation method of the equivalent ITD and 1/3-octave band-based IID factors and their usage to locate a sound source in space. Subjective and objective tests were carried out to examine the effectiveness of the proposed methodology and its applicability to real sound systems. Those experimental results are illustrated in this paper.

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