• Proceedings of the Korean Society of Computer Information Conference
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• 2012.07a
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• pp.345-346
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• 2012

다양한 모바일 디바이스에서는 스피커와 마이크로폰의 자체에서 발생하는 반향(Echo)에 대한 문제점이 제시되어 녹음할 때 생기는 에코를 제거하기 위한 알고리즘 연구의 필요성이 증대되고 있다. 따라서 본 논문은 적응 신호 처리에 기반한 음향 반향 제어 알고리즘을 안드로이드 플랫폼 기반의 JNI를 이용한 설계 방법을 제안한다. 적응 반향 제거 애플리케이션은 일반적인 LMS 적응 필터를 적용하여 사운드에서 반향을 제거하고 FIR 필터를 사용하여 사운드의 웨이브폼에 FIR 필터를 적용한다. 안드로이드 플랫폼 기반의 JNI(Java Native Interface)를 이용한 설계는 기존에 사용되는 소스 코드를 재활용하고 프로그래밍을 쉽게하도록 도와준다.

• Proceedings of the Korea Institute of Convergence Signal Processing
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• 2000.12a
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• pp.185-188
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• 2000

현대의 잡음이 많은 환경에서 적응 음향반향제거기는 배경잡음의 영향으로 원활한 통화환경을 제공할 수 없다. 이러한 문제점을 해결하기 위하여 음향반향 제거와 더불어 배경잡음을 제거하는 결합구조의 적응 음향반향제거기가 제안되었다. 본 논문에서는 기존의 결합구조가 가지는 단점을 보완하여 적응 음향반향제거기의 성능을 향상시켰다. 제안한 결합구조는 적응 음향잡음제거기의 기준입력 신호를 적응 음향잡음제거기의 오차신호와 같게 구성함으로서 배경잡음 신호뿐만 아니라 잔여반향 신호도 효율적으로 제거할 수 있다. 성능 평가를 위한 실험결과, 제안한 방법이 기존의 방법에 비하여 ERLE 성능이 수렴 구간에서 3㏈ 이상 향상되었음을 확인하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.05a
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• pp.839-842
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• 2010

In this paper, we propose a residual echo and noise reduction scheme for hands-free telephony applications. The proposed algorithm uses a noise robust modified AP algorithm which estimate well echo path in noisy and whitens residual echo signal using linear prediction at non double-talk duration. It is confirmed that the proposed algorithm shows better performance from acoustic interference cancellation (AIC) viewpoint.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.36 no.5
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• pp.353-360
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• 2017

The single sensor based acoustic reflection control system separates the incident and reflected signals from the single sensor output, and reduces the reflected signal by generating an out-of-phase signal from the incident signal component. In this paper, we propose an optimal filter design method for a single sensor based reflection control system. In the proposed method, it is shown that the optimum control filter design is possible by using the measured impulse responses of the reflection and control paths. The reflection control algorithm based on the proposed optimal filter achieves better performance than the conventional adaptive filter-based algorithm and effectively controls the reflection without the initial convergence time. We performed computer simulations using the signals obtained in a 1-dimensional acoustic duct environment, and from the simulation results, it was confirmed that the proposed optimal filter has robust performance even in noisy environment.

• Journal of Korea Multimedia Society
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• v.3 no.3
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• pp.263-271
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• 2000

In this paper, an acoustic echo canceller(AEC) is implemented by ADSP-2181. This AEC uses a noise robust adaptive algorithm and a postprocessing method which attenuates residual echo using cross-correlation between estimated error signal and microphone input signal. We propose new postprocessing method that uses two thresholds to prevent signal distortion after postprocessing and to improve the performance of AEC without extra computational burden. Through experiments using a 16 bit fixed-point DSP board (ADSP-2181 EZ-KIT Lite board), it is shown that the noise robust adaptive algorithm performs well in the double-talk situations and the convergence speed is comparable to NLMS. Using the postprocessor, ERLE is improved about 20 dB. As a result, the AEC with a postprocessor shows better performance than conventional ones.

• 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.

• 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.

• The Journal of the Acoustical Society of Korea
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• v.36 no.1
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• pp.57-63
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• 2017

This paper presents an active acoustic-reflection control algorithm based on a single sensor. The proposed algorithm operates in a system comprising a single sensor located nearby the reflective surface and a control transducer mounted on the reflective surface. First, the incident and reflected acoustic signals are separated from the sensor signal, and a control signal is generated using the separated signals. For the signal separation, the proposed algorithm requires the response of the reflection path which is estimated from the acoustic response between an external sound source and the sensor. Finally, the control filter is adjusted using the FxLMS (Filtered-x Least Mean Square) algorithm. To verify the effectiveness of the proposed algorithm, it was implemented in real time using a DSP (Digital Signal Processing) board, and the experimental results obtained in one-dimensional air-acoustic environment show that the reflections of the 1 kHz burst can be reduced by 11.6 dB.

• Journal of the Korean Institute of Telematics and Electronics S
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• v.35S no.9
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• pp.104-110
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• 1998

This paper proposes a architecture of the nonlinear echo canceller for the full-duplex transmission, which is easy to implement. The architecture is designed to cancel the nonlinear echo signals due to the hybrid or the data converter, as well as the linear echo signals, by adding a small number of taps to the FIR structure for the linear echo cancellation. The number of tap to be add depends on how much the linear echo signals are cancelled. A computer program for this architecture was written and run. Numerical results of the computer simulations show good performances on cancellations of the nonlinear echo signals by adding a small number of taps.