• Title/Summary/Keyword: 빔포밍 마이크

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A study on implementation of beam forming system for LED communication using micro controller (마이크로 컨트롤러를 이용한 LED 통신의 선택적 빔 포밍 시스템 구현에 관한 연구)

  • Lee, JungHoon;Kim, Chan;Cha, Jaesang
    • Journal of Satellite, Information and Communications
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    • v.7 no.2
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    • pp.25-29
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    • 2012
  • In this paper, we implemented LED beam forming communication system controlled by stepping motor. ATMega1284 was used as a MCU of main control board which has two main external IO, one is RS232 for connection with PC, the other is PORT for connection with motor driving board. Stepping motor rotated 360 degree when provided 160 clock and its rotation radius was increased by Archimedian Spiral. So LED can provide its light anywhere in the space and its beam forming was controlled by PC connected with RS232 of main control board. The action of beam forming was verified via actual HW/SW implementation.

A Feedback and Noise Cancellation Algorithm of Hearing Aids Using Adaptive Beamforming Method (적응 빔형성기법을 이용한 보청기의 궤환 및 잡음제거 알고리즘)

  • Lee, Haeng-Woo
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.35 no.1C
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    • pp.96-102
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    • 2010
  • This paper proposes a new adaptive algorithm to cancel the acoustic feedback and noise signals in the digital hearing aids. The proposed algorithm improves its convergence performances by canceling the speech signal from the residual signal using two microphones. The feedback canceller firstly cancels the feedback signal among the mic signal, and then it is reduced the noise using the beamforming method. To verify the performances of the proposed algorithm, the simulations were carried out for some cases. As the results of simulations, it was proved that the feedback canceller and the noise canceller advance about 14.43 dB for SFR, 10.19 dB for SNR respectively during speech, in the case of using the new algorithm.

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.

A method of wall absorption treatment for enhancing the speech intelligibility at a directional microphone array in a room (실내 공간 내 지향성 마이크 어레이에서의 음성 명료도 개선을 위한 벽면 흡음 처리 방법)

  • Ko, Byeong-Yun;Ih, Jeong-Guon;Cho, Wan-Ho
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.6
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    • pp.649-659
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    • 2021
  • Wall absorption treatment effectively reduces reverberation, but requires a large area for a live room and each wall absorption affects speech intelligibility differently. In this study, we try to find the most effective wall for the absorption treatment using the beamforming array microphone in terms of speech intelligibility. The absorption importance factor is defined by using the collision number of reflected sounds on each wall. It allows estimating how much the speech signal will be enhanced by the absorption treatment. A cuboid room with a size of 107 m3 and a reverberation time of 1.1 s is selected for the simulation. When a Helmholtz-type absorption is treated on the wall with the most significant importance factor, the modified clarity for 500 and 1k Hz is improved by 5.1 dB and 4.8 dB respectively, and the speech transmission index is enhanced by 0.06. The difference in results between the proposed method and commercial simulation code is less than a Just-Noticeable Difference (JND). The absorption treatment on the wall with the most significant importance factor shows improvement greater than the wall with the largest area, and its difference is larger than a JND value.

Wide Coverage Microphone System for Lecture Using Ceiling-Mounted Array Structure (천정형 배열 마이크를 이용한 강의용 광역 마이크 시스템)

  • Oh, Woojin
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.22 no.4
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    • pp.624-633
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    • 2018
  • While the multimedia lecture system has been getting smart using immerging technology, the microphone still relies on the classical approach such as holding in hand or attaching on the body. In this paper, we propose a ceiling mounted array microphone system that allows a wide reception coverage and instructors to move freely without attaching microphone. The proposed system adopts cell and handover of mobile communication instead of a complicated beamforming method and implements a wide range microphone over several cells with low cost. Since the characteristics of unvoiced speech is similar to Pseudo Noise it is shown that soft handover are possible with 3 microphones connected to delay-sum multipath receiver. The proposed system is tested in $6.3{\times}1.5m$ area. For real-time processing the correlation range can be reduced by 82% or more, and the output latency delay can be improved by using the delay adaptive filter.

Multi-channel input-based non-stationary noise cenceller for mobile devices (이동형 단말기를 위한 다채널 입력 기반 비정상성 잡음 제거기)

  • Jeong, Sang-Bae;Lee, Sung-Doke
    • Journal of the Korean Institute of Intelligent Systems
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    • v.17 no.7
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    • pp.945-951
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    • 2007
  • Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.