• Title/Summary/Keyword: Acoustic Problem

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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.

A study on the robust speaker recognition algorithm in noise surroundings (주변 잡음 환경에 강한 화자인식 알고리즘 연구)

  • Jung Jong-Soon
    • Journal of the Korea Society of Computer and Information
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    • v.10 no.6 s.38
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    • pp.47-54
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    • 2005
  • In the most of speaker recognition system, speaker's characteristics is extracted from acoustic parameter by speech analysis and we make speaker's reference pattern. Parameters used in speaker recognition system are desirable expressing speaker's characteristics fully and being a few difference whenever it is spoken. Therefore we su99est following to solve this problem. This paper is proposed to use strong spectrum characteristic in non-noise circumstance and prosodic information in noise circumstance. In a stage of making code book, we make the number of data we need to combine spectrum characteristic and Prosodic information. We decide acceptance or rejection comparing test pattern and each model distance. As a result, we obtained more improved recognition rate than we use spectrum and prosodic information especially we obtained stational recognition rate in noise circumstance.

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Low-Frequency Normal Mode Reverberation Model (저주파수 정상모드 잔향음 모델)

  • Oh, Suntaek;Cho, Sungho;Kang, Donhyug;Park, Kyoungju
    • The Journal of the Acoustical Society of Korea
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    • v.34 no.3
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    • pp.184-191
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    • 2015
  • In this paper, a normal mode reverberation model for a range-independent environment of shallow water is proposed to calculate the reverberation level in the low-frequency range. Normal mode is used to calculate the acoustic energy propagating from the source to the scattering area and from the scattering area to the receiver. Each mode is decomposed into up and down going waves to consider scattering strength at the scattering area. The scattering functional form combines Lambert's law with a Gaussian-like term near the specular direction based on Kirchhoff approximation considering bottom condition. For verification of the suggested model, the result is relatively compared to several solutions of the problem XI and XV in the Reverberation Modeling Workshop I sponsored by the US Office of Naval Research.

Signal Processing for Speech Recognition in Noisy Environment (잡음 환경에서 음성 인식을 위한 신호처리)

  • Kim, Weon-Goo;Lim, Yong-Hoon;Cha, Il-Whan;Youn, Dae-Hee
    • The Journal of the Acoustical Society of Korea
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    • v.11 no.2
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    • pp.73-84
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    • 1992
  • This paper studies noise subtraction methods and distance measures for speech recognition in a noisy environment, and investigates noise robustness of the distance measures applied to the problem of isolated word recognition in white Gaussian and colored noise (vehicle noise) environments. Noise subtraction methods which can be used as a pre-processor for the speech recognition system, such as the spectral subtraction method, autocorrelation subtraction method, adaptive noise cancellation and acoustic beamforming are studied, and distance measures such and Log Likelihood Ratio ($d_{LLR}$), cepstral distance measure ($d_{CEP}$), weighted cepstral distance measure ($d_{WCEP}$), spectral slope distance measure ($d_{RPS}$) and cepstral projection distance measure ($d_{CP},\;d_{BCP},\;d_{WCP},\;d_{BWCP}$) are also investigated. Testing of the distance measures for speaker-dependent isolated word recognition in a noisy environment indicate that $d_{RPS}\;and\;d_{WCEP}$ which weigh higher order cepstral coefficients more heavily give considerable performance improvement over $d_{CEP}and\;d_{LLR}$. In addition, when no pre-emphasis is performed, the recognizer can maintain higher performance under high noise conditions.

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Design and Implementation of MAC Protocol for Underwater Mobile Ad-hoc Networks (수중 모바일 애드 혹 네트워크를 위한 MAC 프로토콜 설계 및 구현)

  • Lee, Jin-Young;Yun, Nam-Yeol;Park, Soo-Hyun
    • Journal of the Institute of Electronics and Information Engineers
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    • v.51 no.4
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    • pp.76-89
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    • 2014
  • In this paper, we propose a MAC(Media Access Control) protocol based on flexible RWT(RTS Waiting Time) for underwater mobile ad-hoc networks with a three-way handshaking mechanism. This protocol can solve a problem of collision between RTS(Request-To-Send) and CTS(Clear-To-Send) packets in existing MACA(Multiple Access with Collision Avoidance) protocol. This proposed MAC protocol is also an effective protocol which can apply to underwater mobile ad-hoc networks in a real field by using implementable technologies. We set flexible RTS Waiting Time called RWT, considering various characteristics of underwater environment. It is possible to support variable network size according to node mobility. Finally, we conduct a performance evaluation between proposed MAC protocol and existing MACA based MAC protocol through practical implementation and experiment. As a result, we verify the superiority of our proposed MAC protocol in terms of throughput, packet drop rate, average transmission time, energy consumption and channel utilization.

Parametric Array Sonar System Based on Maximum Likelihood Detection (최대우도 검파에 기반한 파라메트릭 어레이 소나 시스템)

  • Han, Jeong-Hee;Lee, Chong-Hyun;Paeng, Dong-Guk;Bae, Jin-Ho;Kim, Won-Ho
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.48 no.1
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    • pp.25-31
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    • 2011
  • In the underwater communications, transmitted acoustic signal is corrupted by interference from multipath. A parametric array transducer is capable of radiating a narrow beam with very low sidelobe levels. In certain cases, the parametric array transducer can help the multipath problem. To improve the performance of the underwater communications, the statistical signal processing methods will be required. In the paper, the communication system using a parametric array transducer was demonstrated. To detect the received signal of the communication system based on the on-off keying, the maximum likelihood method using averaged signal for a particular window size is used. The communication system has GUI using LebVIEW which allows the user to change the parameter. The GUI can also be easily modified based on the characteristics of a parametric array transducer. The implemented system can effectively evaluate the performance of the parametric array transducer.

Optimal Muffler Design Considering the Insertion Loss Calculated Outside the Duct (덕트 밖에서 계산된 삽입 손실을 고려한 머플러 최적 설계)

  • Lee, Jong Kyeom;Oh, Kee Seung;Lee, Jin Woo
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.40 no.5
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    • pp.497-503
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    • 2016
  • In this study, we formulate an acoustical topology optimization problem to optimally design a partition layout inside the expansion chamber of a muffler. The lower-limit insertion loss value at a target frequency is constrained, and the partition volume is selected as an object function. In this study, we calculate the insertion loss outside the duct, while to determine the noise-attenuation performance, we use the insertion loss value calculated inside the duct or transmission loss value obtained in a previous study. We employ the finite-element model for acoustical analysis, and we determine the transmission of an incident acoustic wave through each finite element using the functions of design variables that change continuously between "0" and "1." The rigid body elements, which totally reflect incident waves, build up partitions. Finally, we compare optimal topologies that depend on the target frequency and the allowed lower-limit value of insertion loss.

An Array Beampattern Synthesis Using Adaptive Array Method and Partial Constrained Adaptation (최소 자승 평균오차와 부분 적응을 사용한 배열 빔 형성기법)

  • Lim Jun-Seok;Choi Nakjin;Sung Koeng-Mo;Kim Hyun-Seok
    • The Journal of the Acoustical Society of Korea
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    • v.23 no.8
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    • pp.570-575
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    • 2004
  • In the underwater acoustic systems. we can receive signals and retrieve information about a target by using a beamforming method. The most important thing in the beamforming is finding the way to optimize the mainlobe beamwidth and the sidelobe level to the desired value. One of the prominent results of beamforming method. which has been studied. is Philip's weighting function method(1) . Philip's method adaptively adjusts its weights of array to meet the desired mainlobe beamwidth and sidelobe level. It is very similar to the design method in adaptive filter. However. this method cannot easily bring us to the desired sidelobe level due to complementary relation between mainlobe beamwidth and sidelobe level. In this paper, we propose a new algorithm using partial constrained adaptation. This method makes us circumvent the above problem and meet the specification of design easily. The proposed algorithm presents a Pattern synthesis that designer can easily control the mainlobe beamwidth and the sidelobe level to the desired value while calculation time to converge is decreasing.

Structural health monitoring of high-speed railway tracks using diffuse ultrasonic wave-based condition contrast: theory and validation

  • Wang, Kai;Cao, Wuxiong;Su, Zhongqing;Wang, Pengxiang;Zhang, Xiongjie;Chen, Lijun;Guan, Ruiqi;Lu, Ye
    • Smart Structures and Systems
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    • v.26 no.2
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    • pp.227-239
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    • 2020
  • Despite proven effectiveness and accuracy in laboratories, the existing damage assessment based on guided ultrasonic waves (GUWs) or acoustic emission (AE) confronts challenges when extended to real-world structural health monitoring (SHM) for railway tracks. Central to the concerns are the extremely complex signal appearance due to highly dispersive and multimodal wave features, restriction on transducer installations, and severe contaminations of ambient noise. It remains a critical yet unsolved problem along with recent attempts to implement SHM in bourgeoning high-speed railway (HSR). By leveraging authors' continued endeavours, an SHM framework, based on actively generated diffuse ultrasonic waves (DUWs) and a benchmark-free condition contrast algorithm, has been developed and deployed via an all-in-one SHM system. Miniaturized lead zirconate titanate (PZT) wafers are utilized to generate and acquire DUWs in long-range railway tracks. Fatigue cracks in the tracks show unique contact behaviours under different conditions of external loads and further disturb DUW propagation. By contrast DUW propagation traits, fatigue cracks in railway tracks can be characterised quantitatively and the holistic health status of the tracks can be evaluated in a real-time manner. Compared with GUW- or AE-based methods, the DUW-driven inspection philosophy exhibits immunity to ambient noise and measurement uncertainty, less dependence on baseline signals, use of significantly reduced number of transducers, and high robustness in atrocious engineering conditions. Conformance tests are performed on HSR tracks, in which the evolution of fatigue damage is monitored continuously and quantitatively, demonstrating effectiveness, adaptability, reliability and robustness of DUW-driven SHM towards HSR applications.

An Experimental Study of Radiated So from Elastic Thin Plate in a Turbulent Boundary Layer (난류 유동장 내에 놓인 탄성을 갖는 박판의 방사소음에 대한 실험적 연구)

  • Lee, Seung-Bae;Gwon, O-Seop;Lee, Chang-Jun
    • Transactions of the Korean Society of Mechanical Engineers B
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    • v.25 no.10
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    • pp.1327-1336
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    • 2001
  • The structural modes driven by the low wave-number components of smooth elastic wall pressure provide a relatively weak coupling between the flow and the wall motion. If the elastic thin plate has any resonant mode whose wave-number of resonance coincides with $\omega$/U$\sub$c/, the power will be transmitted to those modes of vibration by the flows. We examine the problem in which the elastic thin plate is subject to pressure fluctuations under turbulent boundary layer. Measurements are presented of the frequency spectra of the near- and far-field pressures and radiated sound contributed by the various wave modes of the thin elastic plate. Dispersion equation for wave motions of elastic plate is used to investigate the effect of bending waves of relatively low wave number on radiated sound. The low wave-number motion of elastic plate is observed to have much less influence on the low-frequency energy of wall pressure fluctuations than that of the rediated sound. High amplitude events of the wall pressure are observed to weakly couple with high-frequency energy of radiated sound for case of low tension applied to the plate. The sound source localization is applied to the measurement of radiated sound by using acoustic mirror system.