• 한국정보통신학회논문지
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• 제11권5호
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• pp.961-968
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• 2007

오늘날, 우리는 어디엔가 엔제나 무전기 통신 장치를 사용할수 있다. 때때로, 우리는 음향잡음환경에서 장치를 사용하였다. 그 음향잡음은 통신장치에서 많은 문제를 만들었다. 음향잡음환경에서는, 말은 음성신호와 잡음신호 양쪽에 신호를 포함하고, 받았기 때문에 깨끗한 정보를 받기위해 보낼수가 없었다. 디지털필터는 바라는 신호를 얻기 위해 옮기는 잡음으로서 유용하였다. 방법의 하나는 자동적으로 맞추는 필터 파라미터로서 적응 잡음 망상조직으로 적응디지털필터를 사용하는 것이다. 본 논문은 두 적응필터 방법에 의하여 현실에서 음향잡음으로서 명료도 알고리즘의 번지라고 할 수가 있다. 하나는 두 입력 채널과 함께 적응잡음 망상조직이라 할 수 있고, 또 다른 것은 하나 입력 채널과 함께 스펙트럼 빼기필터이다. 이 실험의 결과는 제안된 필터로부터 스펙트럼 진폭필터는 움직이지 않는 잡음은 효력이 있는 동안 움직이는 것을 줄이기 위해 사용되어지는 것은 적응잡음망상조직으로 보여준다.

• 한국해양공학회:학술대회논문집
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• 한국해양공학회 2004년도 학술대회지
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• pp.386-389
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• 2004

Beamforming for underwater acoustic communication is affected by the broadband feature of underwater acoustic communication signal, which has the low center frequency compared to the signal bandwidth. In this paper, the baseband equivalent array signal model is derived and we present computer simulation results for the broadband finite impulse response (FIR) beamformer performance according to the FIR filter order and the tap spacing. If the FIR filter order is increased above the optimum value, the beamformer performance is degraded. Also the tap spacing is related to the optimum FIR filter order.

• ETRI Journal
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• 제32권6호
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• pp.921-931
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• 2010

It has been shown that the principal subspace-based multichannel Wiener filter (MWF) provides better performance than the conventional MWF for suppressing interference in the case of a single target source. It can efficiently estimate the target speech component in the principal subspace which estimates the acoustic transfer function up to a scaling factor. However, as the input signal-to-interference ratio (SIR) becomes lower, larger errors are incurred in the estimation of the acoustic transfer function by the principal subspace method, degrading the performance in interference suppression. In order to alleviate this problem, a principal subspace modification method was proposed in previous work. The principal subspace modification reduces the estimation error of the acoustic transfer function vector at low SIRs. In this work, a frequency-band dependent interpolation technique is further employed for the principal subspace modification. The speech recognition test is also conducted using the Sphinx-4 system and demonstrates the practical usefulness of the proposed method as a front processing for the speech recognizer in a distant-talking and interferer-present environment.

• 음성과학
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• 제9권3호
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• pp.17-24
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• 2002

The goal of this research is the real-time implementation of an AEC (Acoustic Echo Canceller) using the floating-point digital signal processor of TMS320C31. We employ an FIR-type adaptive filter with the conventional NLMS (Normalized Least Mean Square) algorithm for the adaptation of filter coefficients. We program and optimize the system in the assembler level to make it run in real-time. With 8 kHz sampling rate, the implemented AEC requires $46\;\mu$sec and $77\;\mu$sec computational time per sample for 128-and 256-tap filter, respectively. It corresponds to 37% and 62% of maximum computational ability of TMS320C31 DSP.

• 한국수산과학회지
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• 제23권6호
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• pp.451-456
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• 1990

Underwater acoustic navigation and position fixing systems have been extensively used not only in surface position fixing but also in underwater position fixing. Tn recently, application of these systems has been in the field of underwater inspection of offshore platforms, where it is vital to track the position of an unmanned submersible or diver carrying underwater cameras and nondestructive testing equipment. But these systems are included the fixing errors as results of a signal with additive noise, the attenuation of sound and the interference effects due to multipath reflection and forward scattering. In this paper to improve the position fixing by the supershort baseline acoustic position system, a method to apply the Kalman filter to the fix of the system is proposed and the digital simulation under noise condition is conducted. The optimal positions by the Kalman filter are compared with original positions, and it is confirmed that the results of the pro-posed method are evidently more accurate.

• 한국해양공학회:학술대회논문집
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• 한국해양공학회 2000년도 추계학술대회 논문집
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• pp.222-227
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• 2000

"Side scan sonar" using acoustic signal has been developed to survey cable laying, sunken bodie\ulcorner bottom and so on. It use the acoustic signals, which are emitted from two transducer arrays, to get gemetri\ulcorner target area. This system consists of transceiver board, towed body, and deck unit. The transceiver board, w\ulcorner watertight canister of the towed body, controls the transmitting and receiving of 400kHz acoustic signals from \ulcorner After receiving the scattered signals, it processes the filtering, AGF(Automatic Gain Control), TVG(Time Heterodyne. The deck unit is composed of the signal processing part, A/D converter, power supplier, and real\ulcorner And the towed body has been designed to satisfy the optimal hydrodynamic behavior during towing. The de\ulcorner theory of transceiving part and some results from field-experiments will be introduced here.

• 한국소음진동공학회:학술대회논문집
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• 한국소음진동공학회 2001년도 추계학술대회논문집 II
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• pp.1104-1108
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• 2001

It is a difficult and important task to measure the reverberation time of an acoustic room with a short reverberation time. This paper presents a new technique to measure the reverberation time of an acoustic room with low value of BT60. The digital signal processing technique used to do this is the wavelet filter which is very flexible to design the 1/n octave band filter and has no delay problem compared with the conventional IIR digital filter. This method is successfully applied to the measurement of the reverberation time at low frequency band of famous concert halls in Korea.

• 한국음향학회지
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• 제21권5호
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• pp.438-445
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• 2002

본 논문에서는 격자 트랜스버설 결합 (LTJ) 적응필터를 시변 변환영역 적응필터 관점에서 해석함으로써 필터계수보상의 필요성을 보다 쉽고 일반적으로 해석하였다. 또한, 음성 신호가 단구간에서 정적이라는 특성을 이용하여 필터계수 보상을 위한 계산량을 감소시킬 수 있는 방법을 제안하였으며, 모의 음성신호와 실제 음성신호를 사용한 실험을 통하여 효용성을 입증하였다. 제안된 적응필터는 필터계수 보상을 위한 계산량이 95% 감소되었으며, 1000탭을 사용하는 음향반향제거기의 경우 전체 시스템의 계산량을 82% 감소시킬 수 있다.

• 전자공학회논문지B
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• 제32B권11호
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• pp.1473-1480
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• 1995

When the LMS algorithm is employed in the transversal filter structure, the computational complexity can be kept reasonably low. However, if the impulse response to be estimated is very long or signals involved are highly correlated like a speech the convergence speed becomes slow. The lattice filter is an excellent alternative to improve convergence speed since the lattice structure inherently has the orthogonal property among the backward prediction errors, but at the expense of the excessive computational load. If the input signal to be used can be sufficiently well modeled as a .RHO.-th order autoregressive(AR) process, the reflection coefficients after the .RHO.- th stage will be close to zero. Then, instead of employing the full lattice structure, the joint lattice filter structure can be implemented in conjunction with the transversal filter structure after the .RHO.-th stage. We propose, in this paper, this new lattice/transversal joint structure, and we will call it the reduced lattice filter. Using the reduced lattice filter, we are now able to achieve the performance as good as that of the lattice filter, while maintaining the complexity as low as that of the transversal filter. The proposed filter is particularly useful for an acoustic echo canceller due to the highly correlatedness nature of speeches and the long and frequently changing echo paths.

• 로봇학회논문지
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• 제12권4호
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• pp.402-410
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• 2017

Acoustic based localization is essential to operate autonomous robotic systems in underwater environment where the use of sensorial data is limited. This paper proposes a localization method using artificial underwater acoustic sources. The proposed method acquires directional angles of acoustic sources using time difference of arrivals of two hydrophones. For this purpose, a probabilistic approach is used for accurate estimation of the time delay. Then, Gaussian sum filter based SLAM technique is used to localize both acoustic sources and underwater vehicle. It is performed by using bearing of acoustic sources as measurement and inertial sensors as prediction model. The proposed method can handle directional ambiguity of time difference based source localization by generating Gaussian models corresponding to possible locations of both front and back sides. Through these processes, the proposed method can provide reliable localization method for underwater vehicles without any prior information of source locations. The performance of the proposed method is verified by experimental results conducted in a real sea environment.