• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.6 no.1
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• pp.23-28
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• 2013

In this paper, background noise reduction method using dual microphone is proposed in mobile environment. Each Signal, reference and primary, would be replaced by microphone input signals, which were measured by reference and primary microphones, and then, noise reduction was performed using FDAF. After then, residual and background noise would be estimated and reduced by MMSE-LSA. For consistent noise reduction performance, result of VAD that could be caculated by PLD between two microphones was used.

• The Transactions of the Korean Institute of Electrical Engineers C
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• v.55 no.5
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• pp.263-266
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• 2006

An optical microphone was developed using a dual-core multi-mode fiber block and a membrane type micromirror. The fiber block serves as a compact optical head, and the micromirror as a reflective diaphragm. The micromirror is designed to be suspended through a silicon bar connected t a frame, allowing for displacement induced by acoustic waves. The optical head is implemented by integrating two multi-mode fibers in a single block, and used to transfer light signals between it and the diaphragm. For the assembled microphone, its static characteristics were observed to reveal the operating point defined as the optimum distance between the optical head and the diaphragm. And its dynamic response was tested to exhibit a frequency bandwidth of 3 kHz with the variation of $\sim5dB$.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.26 no.5
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• pp.808-811
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• 2022

In this paper, a wide-range microphone system for lectures using dual 3D sensors is proposed. A previous work using a single sensor had lowering the detecting threshold to support wide-area. However it was found that an error occurred when lecturer wears clothes with low reflectivity or has small body size. When multiple sensors are used to expand the coverage it could be cause various problems. Each sensor could show different distance to the same target. We derive the rotation angle and and compensate for lecturing microphone system using sensors on the line. The proposed method shows a little improvement in performance by about 1dB compared to the previous works but the performance is uniform in all areas regardless of reflectivity.

• Korean Journal of Optics and Photonics
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• v.17 no.1
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• pp.94-98
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• 2006

An optical microphone based on a dual-core fiber collimator and a membrane type micromirror serving as an optical head and a reflective diaphragm respectively was implemented. The micromirror diaphragm is suspended by a thin silicon bar linked with a frame, thus it is subject to a displacement induced by acoustic waves. The optical head incorporating two collimators integrated in a single housing gives light to and receives it from the diaphragm, rendering the optical microphone structure simple and compact. This dual-core collimator having a slowing varying beam profile facilitates the initial alignment of the optical head with the diaphragm, especially the distance between them. For the assembled microphone, the static characteristics were investigated tofind the operation point defined as the optimum distance between the head and the diaphragm, and a frequency response with a variation of about $\pm$5 dB for the range of up to 3kHz was achieved.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.2
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• pp.123-129
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• 2017

This paper proposes a new nonnegative matrix factorization (NMF) based direction-of-arrival (DOA) estimation method for multiple sound sources using a dual microphone array. First of all, sound signals coming from the dual microphone array are segmented into consecutive analysis frames, and a steered-response power phase transform (SRP-PHAT) beamformer is applied to each frame so that stereo signals of each frame are represented in a time-direction domain. The time-direction outputs of SRP-PHAT are stored for a pre-defined number of frames, which is referred to as a time-direction block. Next, In order to estimate DOAs robust to noise, each time-direction block is normalized along the time by using a block subtraction technique. After that, an unsupervised NMF method is applied to the normalized time-direction block in order to cluster the directions of each sound source in a multiple sound source environments. In particular, the activation and basis matrices are used to estimate the number of sound sources and their DOAs, respectively. The DOA estimation performance of the proposed method is evaluated by measuring a mean absolute error (MAE) and the standard deviation of errors between the oracle and estimated DOAs under a three source condition, where the sources are located in [$-35{\circ}$, 5m], [$12{\circ}$, 4m], and [$38{\circ}$, 4.m] from the dual microphone array. It is shown from the experiment that the proposed method could relatively reduce MAE by 56.83%, compared to a conventional SRP-PHAT based DOA estimation method.

• Journal of Sensor Science and Technology
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• v.21 no.2
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• pp.144-150
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• 2012

A microphone for fully implantable hearing device was generally implanted under the skin of the temporal bone. So, the implanted microphone's characteristics can be affected by the accompanying noise due to masticatory movement. In this paper, the implantable microphone with 2-channels structure was designed for reduction of the generated noise signal by masticatory movement. And an experimental model for generation of the noise by masticatory movement was developed with considering the characteristics of human temporal bone and skin. Using the model, the speech signal by a speaker and the artificial noise by a vibrator were supplied simultaneously into the experimental model, the electrical signals were measured at the proposed microphone. The collected signals were processed using a general adaptive filter with least mean square(LMS) algorithm. To confirm performance of the proposed methods, the correlation coefficient and the signal to noise ratio(SNR) before and after the signal processing were calculated. Finally, the results were compared each other.

• The Journal of the Korea institute of electronic communication sciences
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• v.13 no.5
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• pp.949-958
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• 2018

A voice recognition control system is a system for controlling a peripheral device by recognizing a voice. Recently, a voice recognition control system have been applied not only to smart devices but also to various environments ranging from IoT(: Internet of Things), robots, and vehicles. In such a voice recognition control system, the recognition rate is lowered due to the ambient noise in addition to the voice of the user. In this paper, we propose a dual channel acoustic beamforming hardware architecture for MEMS(: Microelectromechanical Systems) microphones to eliminate ambient noise in addition to user's voice. And the proposed hardware architecture is designed as ASIC(: Application-Specific Integrated Circuit) using TowerJazz $0.18{\mu}m$ CMOS(: Complementary Metal-Oxide Semiconductor) technology. The designed dual channel acoustic beamforming ASIC has a die size of $48mm^2$, and the directivity index of the user's voice were measured to be 4.233㏈.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2000.10a
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• pp.172-175
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• 2000

본 논문에서는 무선마이크용 특정소출력무선기기의 송수신단을 설계하고 제작하여 특성을 측정하였다. 무선마이크용 특정소출력무선기기의 주파수 대역 740.000~752.000MHz를 사용하였고, 점유주락수대폭의 허용치 200kHz, 최대주파수편이 $\pm$75kHz를 만족하도록 설계하였다. PLL Synthesized, 주파수 변조, Dual Conversion 방식을 사용하였으며 Compander Type은 2:1이다. 설계된 무선마이크 시스템의 출력은 10mW이하이고, 50-20,000Hz의 주파수 응답을 가지며 Hi-fi Audio를 전송할 수 있다. 송수신단의 Total Harmonic Distortion 은 1.5% 이하이다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2008.05a
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• pp.147-150
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• 2008

The explanation of Tourism Guide of present whole country main tourist resort helps to understand the tourist resort. However, activity space of Tourism Guide is not state that can be understood all by upbringing by a natural voice. Because action of Tourism Guide is much in case of most sightseeing explanation to use microphone and speaker etc., as sticking that attach and uses to clothing and so on uses, there are much vexatious. Treatise that see hereupon makes use of establishment style fixing microphone and embody inputted obscene sounds by On-board system inflecting MCU (ATmega128), MSM7731-02 Oki-Dual Codec to minimize noise using ecad filter, and embodied a control program by serial communication method with filter codec. The resultant audible direction the maximum 59ms, the line echo maximum 27ms, the echo decrease maximum 35dB, it embodied the system which removes the adaptation elder brother noise of the back.

• Journal of the Institute of Electronics and Information Engineers
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• v.54 no.1
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• pp.121-128
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• 2017

In this paper, we propose a dual-channel acoustic event detection (AED) method using nonnegative tensor factorization (NTF) and hidden Markov model (HMM) in order to improve detection accuracy of AED in multisource environments. The proposed method first detects multiple acoustic events by utilizing channel gains obtained from the NTF technique applied to dual-channel input signals. After that, an HMM-based likelihood ratio test is carried out to verify the detected events by using channel gains. The detection accuracy of the proposed method is measured by F-measures under 9 different multisource conditions. Then, it is also compared with those of conventional AED methods such as Gaussian mixture model and nonnegative matrix factorization. It is shown from the experiments that the proposed method outperforms the convectional methods under all the multisource conditions.