• Title/Summary/Keyword: Acoustic signals

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A Performance Analysis of Active Mount with Moving-Coil type Electromagnetic Actuator Installed on the Elastic Foundation (탄성지지된 가동코일형 능동마운트의 성능 분석)

  • Jung, Woo-Jin;Bae, Soo-Ryong;Jeon, Jae-Jin
    • Proceedings of the Korean Society for Noise and Vibration Engineering Conference
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    • 2014.10a
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    • pp.227-231
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    • 2014
  • Underwater radiated noise is one of the vital factors in underwater weapon systems like submarine. A passive elastic mount is an effective reduction method for the vibration from a ship-board machinery transmitted to the hull which is radiated as noise outside the hull. A passive elastic mount shows the limitation on the vibration reduction and needs multi stage mounting system including double stage one to meet the required underwater radiated noise criteria. It is necessary for the multi stage mounting system to consider the large displacement in the underwater shock explosion. So it is difficult to apply the multi stage mounting system in submarine because of space limitation. Also recent navy sonar system are trying to have the capability to detect the ship-borne acoustic signals in the low frequency range. An alternative to the passive mount is an active mount with moving-coil type electromagnetic actuator based on a conventional rubber mount in the low frequency range. In the previous paper, 4 active mounts with moving-coil type electromagnetic actuator based on the rubber mount were installed on the hard floor of the facility, which means no consideration on the elastic foundation effect for the control of the active mounts was taken into account. In this study, an experimental performance analysis on the active mount was carried out using 4 active mounts installed on the cylindrical structure to investigate the elastic foundation effect.

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Speech Quality Estimation Algorithm using a Harmonic Modeling of Reverberant Signals (반향 음성 신호의 하모닉 모델링을 이용한 음질 예측 알고리즘)

  • Yang, Jae-Mo;Kang, Hong-Goo
    • Journal of Broadcast Engineering
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    • v.18 no.6
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    • pp.919-926
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    • 2013
  • The acoustic signal from a distance sound source in an enclosed space often produces reverberant sound that varies depending on room impulse response. The estimation of the level of reverberation or the quality of the observed signal is important because it provides valuable information on the condition of system operating environment. It is also useful for designing a dereverberation system. This paper proposes a speech quality estimation method based on the harmonicity of received signal, a unique characteristic of voiced speech. At first, we show that the harmonic signal modeling to a reverberant signal is reasonable. Then, the ratio between the harmonically modeled signal and the estimated non-harmonic signal is used as a measure of standard room acoustical parameter, which is related to speech clarity. Experimental results show that the proposed method successfully estimates speech quality when the reverberation time varies from 0.2s to 1.0s. Finally, we confirm the superiority of the proposed method in both background noise and reverberant environments.

A Feedback and Noise Cancellation Algorithm of Hearing Aids Using Dual Microphones (이중 마이크를 사용한 보청기의 궤환 및 잡음제거 알고리즘)

  • Lee, Haeng-Woo
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.7C
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    • pp.413-420
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    • 2011
  • This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the binaural hearing aids. The convergence performances of the proposed algorithm are improved by updating coefficients of the feedback canceller after the speech signal is cancelled from the residual signal with dual microphones. The feedback canceller firstly cancels the feedback signal from the microphone signal, and then the noise canceller reduces the noise by the beamforming method. To assure that binaural hearing aids converge stably, the left-sided hearing aid only is converged firstly, next the right-sided hearing aid only is converged. To verify performances of the proposed algorithm, simulations were carried out for a speech. As the results of simulations, it was proved that we can advance 14.43dB SFR(Signal to Feedback Ratio) on the average for the feedback canceller, 10.19dB SNR(Signal to Noise Ratio) improvement on the average for the noise canceller, in case that this algorithm is used.

Modeling and Experimental Verification of Echo Characteristics of 3 Dimensional Underwater Target (3차원 수중 표적의 반향특성 모델링과 실험적 검증)

  • You, Seung-Ki;Kim, Sunhyo;Choi, Jee Woong;Kang, Donhyug;Jeong, Dongmin
    • The Journal of the Acoustical Society of Korea
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    • v.33 no.3
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    • pp.174-183
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    • 2014
  • When a active sonar signal is transmitted and returned back from a target, it has been distorted by various properties of acoustic channel such as multipath arrivals. And signals have been appeared to be different form by target position and attitude. Therefore, we simulated the target echo signal using 3 dimensional target model include reflects target features. In this paper, we develop components form of a simulated target model is made up equally spaced highlight points, and each part of the target consists of shape function. We can simulate a target echo signal and Target strength (TS) according to wave incident angle. To verify, we made small scale target in kit form and we had got underwater target signal for comparing simulation result in water tank.

An Objective Performance Analysis of Crosstalk Cancellation Scheme for Sound Rendering Systems Based on Listener Position Tracking (청취자 위치정보 기반 Sound Rendering 시스템 상호간섭 제거기법의 객관적 성능분석)

  • Lee, Jung-Hyuck;Kim, Yeong-Moon;Yoo, Seung-Soo;Kim, Sun-Yong
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.36 no.2C
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    • pp.112-118
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    • 2011
  • In this paper, we conduct an objective performance analysis of the crosstalk cancellation scheme studied in [11]. While the conventional scheme is only applicable to a listener on the optimal listenable region (sweetspot), the space skew/crosstalk cancellation (SS/CC) scheme in [11] can mitigate crosstalk regardless of the listener's position by using listener position tracking (LPT) system. The SS/CC scheme is composed of two parts: LPT-based SS and CC parts. In this paper, the SS/CC scheme is evaluated by some criteria such as follows: condition number, and the balance characteristic, its root mean square error, and running average.

Estimation of surficial sediment thickness using mid-frequency ocean acoustic bottom reflected signals measured in shallow water off Geoje island (거제 인근해역에서 측정된 중주파수 음향 해저면 반사 신호를 이용한 표층 해저면 두께 추정)

  • Kwon, Hyuckjong;Choi, Jee Woong;Son, Su-Uk;Cho, Sungho;Hahn, Jooyoung;park, Joung-Soo;Park, Kyeongju
    • The Journal of the Acoustical Society of Korea
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    • v.35 no.6
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    • pp.419-426
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    • 2016
  • Measurements of bottom loss as a function of grazing angle (in range of $9{\sim}14^{\circ}$) at a frequency range of 4 ~ 8 kHz were conducted on an experimental site off Geoje island in October 2015. Geoacoustic inversion of the surficial sediment thickness is performed using the arrival time difference between the surficial layer and the sub-bottom layer reflected signal. To invert the thickness of surficial sediment, we used the grain size of $8{\sim}10{\phi}$ obtained by KIGAM (Korea Institute of Geoscience and Mineral Resources). The thickness of the surficial sediment was estimated to be 4 ~ 7 m. Finally, this inversion result was compared with the geoacoustic observation conducted by the KIOST (Korea Institute of Ocean Science & Technology) using sub-bottom profiler.

Analysis of the beam pattern of a thickness shear mode vibrator for vector hydrophones (벡터 하이드로폰을 위한 두께 전단형 진동자의 빔 패턴 해석)

  • Kim, Jungsuk;Kim, Hoeyong;Roh, Yongrae
    • The Journal of the Acoustical Society of Korea
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    • v.36 no.3
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    • pp.158-164
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    • 2017
  • Typical hydrophones in line array sensors for early detection of covert underwater targets can measure only sound-pressure-magnitude with the limitation of being unable to identify the direction of an incoming wave. In this study, a thickness shear mode vibrator was proposed as the main component of an inertia type vector hydrophone to measure both magnitude and direction of acoustic signals from targets. The equation to analyze the output voltage of the vibrator to an external force was derived, and the validity of the equation was verified through finite element analysis of a PMN-PT single crystal vibrator. The analysis results from this study will be utilized in the future for the design of inertia type vector hydrophones made of thickness shear vibrators.

Quality Improvement of Karaoke Mode in SAOC using Cross Prediction based Vocal Estimation Method (교차 예측 기반의 보컬 추정 방법을 이용한 SAOC Karaoke 모드에서의 음질 향상 기법에 대한 연구)

  • Lee, Tung Chin;Park, Young-Cheol;Youn, Dae Hee
    • The Journal of the Acoustical Society of Korea
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    • v.32 no.3
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    • pp.227-236
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    • 2013
  • In this paper, we present a vocal suppression algorithm that can enhance the quality of music signal coded using Spatial Audio Object Coding (SAOC) in Karaoke mode. The residual vocal component in the coded music signal is estimated by using a cross prediction method in which the music signal coded in Karaoke mode is used as the primary input and the vocal signal coded in Solo mode is used as a reference. However, the signals are extracted from the same downmix signal and highly correlated, so that the music signal can be severely damaged by the cross prediction. To prevent this, a psycho-acoustic disturbance rule is proposed, in which the level of disturbance to the reference input of the cross prediction filter is adapted according to the auditory masking property. Objective and subjective test were performed and the results confirm that the proposed algorithm offers improved quality.

Design and Fabrication of Compressive Receiver for RFID Signal Detection (RFID 신호 탐지용 컴프레시브 수신기의 설계 및 제작)

  • Jo, Won-Sang;Park, Dong-Chul
    • The Journal of Korean Institute of Electromagnetic Engineering and Science
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    • v.21 no.3
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    • pp.321-330
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    • 2010
  • In this paper, the theoretical background and the specific implementation method of a compressive receiver for RFID signal detection as well as the design method of DDL(Dispersive Delay Line) and chirp LO are described. DDL, which is one of the main components of the compressive receiver, is designed to have $13{\mu}s$ dispersive delay time and 6 MHz bandwidth using the SAW technique based on $LiNbO_3$ material. The chirp LO is designed using DDS(Direct Digital Synthesizer). Also the compressive receiver is fabricated to be installed into the RFID reader. Test results show the maximum frequency error of 25 kHz for single signal input, the receiver sensitivity of -44 dBm, and the maximum frequency error is 75 kHz for 6 multi-tone input signals. These results indicate that the fabricated compressive receiver is working well even in dense RFID operating environments.

Analysis on the Correlation Coefficient for the Diversity Technique Combined with Beamforming Using Measurement Data in Underwater Channel Environments (수중 채널 환경에서 측정 데이터를 이용한 빔형성기가 결합된 다이버시티의 상관 계수 분석)

  • Kim, Min-Sang;Cho, Dae-Young;Park, Jong-Won;Lim, Yong-Kon;Ko, Hak-Lim
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.37A no.12
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    • pp.1023-1030
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    • 2012
  • The diversity techniques can benefit underwater acoustic communications when the distance between sensors is sufficiently apart, and this leads to the increases in the physical size of the communication system: thus it is very hard to practically use such systems in real-environments. Therefore, in this paper, we have collected data from real underwater cannel environments in order to analyze the usability of diversity combined with beamforming techniques. And we have estimated the fading characteristics from the measurement data, and analyzed the correlation coefficients using the estimated fading characteristics. After analyzing the estimated fading characteristics from the measurements data, we found out that by applying diversity techniques on the output signals from beamformers that perform beamforming from different multipath directions, we can reduce the distance between sensors and at the same time benefit from the diversity gain.