• Title/Summary/Keyword: Acoustic filter(음향 필터)

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Impact of face masks on spectral and cepstral measures of speech: A case study of two Korean voice actors (한국어 스펙트럼과 캡스트럼 측정시 안면마스크의 영향: 남녀 성우 2인 사례 연구)

  • Wonyoung Yang;Miji Kwon
    • The Journal of the Acoustical Society of Korea
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    • v.43 no.4
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    • pp.422-435
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    • 2024
  • This study intended to verify the effects of face masks on the Korean language in terms of acoustic, aerodynamic, and formant parameters. We chose all types of face masks available in Korea based on filter performance and folding type. Two professional voice actors (a male and a female) with more than 20 years of experience who are native Koreans and speak standard Korean participated in this study as speakers of voice data. Face masks attenuated the high-frequency range, resulting in decreased Vowel Space Area (VSA) and Vowel Articulation Index (VAI)scores and an increased Low-to-High spectral ratio (L/H ratio) in all voice samples. This can result in lower speech intelligibility. However, the degree of increment and decrement was based on the voice characteristics. For female speakers, the Speech Level (SL) and Cepstral Peak Prominence (CPP) increased with increasing face mask thickness. In this study, the presence or filter performance of a face mask was found to affect speech acoustic parameters according to the speech characteristics. Face masks provoked vocal effort when the vocal intensity was not sufficiently strong, or the environment had less reverberance. Further research needs to be conducted on the vocal efforts induced by face masks to overcome acoustic modifications when wearing masks.

Design and Calibration of Acousto-Optic Tunable Filter(AOTF) for Near Infrared Spectral Analysis (근적외선 분광 분석을 위한 음향광학변조필터의 설계 및 교정)

  • You, Jang-Woo;Kim, Dae-Suk;Kwak, Yoon-Keun;Kim, Soo-Hyun;Lee, Yun-Woo;Hwang, In-Duk
    • Transactions of the Korean Society of Mechanical Engineers A
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    • v.28 no.11
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    • pp.1697-1702
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    • 2004
  • In this paper, we proposed the design and calibration method for the near infrared Acousto-Optic Tunable Filter (AOTF). The theory and design principles of AOTF for the visible light are well known since I.C.Chang has developed the parallel tangent condition for the non-collinear AOTF. Deflection angle, frequency-wavelength relation, spectral resolution, etc. were calculated based on the theory of AOTF. From this result, important parameters - incident and acoustic angle - to fabricate AOTF were decided. We measured the spectral resolution and the relation between electrical driving frequency and the Optical wavelength of diffracted light to calibrate the near infrared AOTF. About 40 ∼ 80 MHz electrical frequency was required to get 1200 ∼ 2200 nm near infrared light. Spectral resolution was less than 10 nm in the near infrared region.

Tunable Narrow-Bandwidth Optical Filter Based on Acoustically Modulated Fiber Bragg Grating (음파에 의한 광섬유 브래그 격자 변조를 이용한 좁은 선폭의 파장가변 광학필터)

  • 염동일;박희수;김병윤
    • Proceedings of the Optical Society of Korea Conference
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    • 2003.07a
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    • pp.198-199
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    • 2003
  • 광섬유 브래그 격자(Fiber Bragg Grating, FBG)는 좁은 선폭과 적은 삽입손실 등의 뛰어난 스펙트럼 특성을 가지고 있어, 고밀도 파장분할 다중화 방식을 이용한 광통신 시스템과 여러 종류의 광섬유센서에 응용되고 있다. 브래그 격자의 광학적 스펙트럼을 능동적으로 조절하기 위하여, 광섬유에 열이나 스트레인을 인가하거나, 음파와 빛의 결합을 이용한 방법이 제안되었으며, 특히 음향 광학 브래그 격자 변조기(Acousto-Optic Superlattice Modulator)는 종 방향 음파(Longitudinal Acoustic Wave)를 이용하여 파장과 크기의 조절이 가능한 사이드 밴드(Side Band)를 생성해 내었다. (중략)

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Optimum Design and Simulation of SAW Filters for Personal Communication Systems (PCS 이동통신용 SAW필터의 최적화 설계 시뮬레이션)

  • Chung, Yeong-Jee
    • The Journal of the Acoustical Society of Korea
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    • v.16 no.3
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    • pp.86-93
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    • 1997
  • A Design & Simulation Tools of Surface Acoustic Wave(SAW) Filters for Mobile Communication Systems, which is based on Optimization of Impulse Samples with Object Function of Amplitude, Ripple and Group Delay Characteristics, is developed and is also evaluated by designning and simulating the SAW IF Filter for PCS. In Optimization Process, fast calculation algorithm of Object Function is proposed. With this Design Tools, Transversal SAW IF Filters can be easily designed under limited conditions of small chip size and package size. It may be also applicable to wide Band Pass Filters in future Communication Systems such as FPLMTS.

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Performance Improvement of Double-talk Detector Using Normalized Error Signal Power (정규화된 오차신호 전력을 이용한 동시통화 검출기의 성능 개선)

  • Heo, Won-Chul;Bae, Keun-Sung
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.5C
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    • pp.478-486
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    • 2007
  • Double-talk detection errors can result in either large residual echo or distorting the near-end talker's input speech. Thus accurate double-talk detection is an important problem in the acoustic echo canceller to improve the speech quality. In the double-talk detection algorithm using a cross-correlation coefficient, double-talk detection errors can occur in the initial convergence period of an adaptive filter or in noisy environment since the cross-correlation coefficient becomes large in such situations. In this paper, we propose a new double-talk detection algorithm based on the cross-correlation method using a normalized error signal power to reduce the double-talk detection errors. The experimental results have shown the performance improvement of an acoustic echo canceller as well as the noise-robustness of the proposed double-talk detector.

A Study on the fabrication of Bandpass filter Using a Simulator (시뮬레이터를 이용한 대역통과 필터 제작)

  • 유일현
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.2
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    • pp.33-39
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    • 2000
  • We have studied to obtain the frequency characteristics of the Surface Acoustic Wave(SAW) bandpass filter, having low shape factor, it's interdigital transducer(IDT) was formed on the 35° Y-cut X-propagation Quartz substrate and was evaporated by Aluminium. And then, we performed computer-simulation by a simulator. And, we can design that the apodization weighted type IDT as an input transducer of the filter and the withdrawal weighted type IDT as an output transducer of the filter from the results of our computer-simulation. Also, we have employed that the number of pairs of the input and output IDT are 2200 pairs and 1000pairs, respectively and used the Kaiser-Bessel window function in order to minimize the effect of ripple. And, while the width and the space of IDT's finger are 6㎛ m and 5.75㎛, respectively and we could obtain the resonable results when the IDT thickness was 6000Å in consideration of the ratio of SAW's wavelength, and IDT aperture is 2mm. Frequency response of the fabricated SAW bandpass filter has the property that the center frequency is about 70MHz, shape factor is less than 1.3, bandwidth at the 1.5dB is probably 1.3MHz, out-band attenuation is almost -45dB, insertion loss is 19dB and ripple in the width of bandpass is 1dB approximately. Therefore, these frequency characteristics of the fabricated SAW bandpass filter are agreed well with the designed values.

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A Study on the Robust Double Talk Detector for Acoustic Echo Cancellation System (음향반항 제거 시스템을 위한 강인한 동시통화 검출기에 관한 연구)

  • 백수진;박규식
    • The Journal of the Acoustical Society of Korea
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    • v.22 no.2
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    • pp.121-128
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    • 2003
  • Acoustic Echo Cancellation(m) is very active research topic having many applications like teleconference and hands-free communication and it employs Double Talk Detector(DTD) to indicate whether the near-end speaker is active or not. However. the DTD is very sensitive to the variation of acoustical environment and it sometimes provides wrong information about the near-end speaker. In this paper, we are focusing on the development of robust DTD algorithm which is a basic building block for reliable AEC system. The proposed AEC system consists of delayless subband AEC and narrow-band DTD. Delayless subband AEC has proven to have excellent performance of echo cancellation with a low complexity and high convergence speed. In addition, it solves the signal delay problem in the existing subband AEC. On the other hand, the proposed narrowband DTD is operating on low frequency subband. It can take most advantages from the narrow subband such as a low computational complexity due to the down-sampling and the reliable DTD decision making procedure because of the low-frequency nature of the subband signal. From the simulation results of the proposed narrowband DTD and wideband DTD, we confirm that the proposed DTD outperforms the wideband DTD in a sense of removing possible false decision making about the near-end speaker activity.

Design and output control technique of sonar transmitter considering impedance variation of underwater acoustic transducer (수중 음향 트랜스듀서의 임피던스 변화를 고려한 소나 송신기의 설계 및 출력 제어 기법)

  • Shin, Chang-Hyun;Lee, Yoon-Ho;Ahn, Byoung-Sun;Yoon, Hong-Woo;Kwon, Byung-Jin;Kim, Kyung-Seop;Lee, Jeong-Min
    • The Journal of the Acoustical Society of Korea
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    • v.41 no.5
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    • pp.481-491
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    • 2022
  • The active sonar transmission system consists of a transmitter that outputs an electrical signal and an underwater acoustic transducer that converts the amplified electrical signal into an acoustic signal. In general, the transmitter output characteristics are dependent on load impedance, and an underwater acoustic transducer, which is a transmitter load, has a characteristic that the electrical impedance varies largely according to frequency when driven. In such a variable impedance condition, the output of the active sonar transmission system may become unstable. Hence, this paper proposes a design and control technique of a sonar transmitter for transmitting a stable transmission signal even under variable impedance conditions of an underwater acoustic transducer in an active sonar transmission system. The electrical impedance characteristics of the underwater acoustic transducer are experimentally analyzed, and the sonar transmitter is composed of a single-phase full-bridge inverter, an LC filter, and a matching circuit. In this paper, the design and output control method of the sonar transmitter is proposed to protect the transmitter and transducer. It can secure stable output voltage characteristics even if it transmits the Linear Frequency Modulation (LFM) signal. The validity is verified through the simulation and the experiment.

Implementation of Adaptive Feedback Cancellation Algorithm for Multichannel Digital Hearing Aid (다채널 디지털 보청기에 적용 가능한 Adaptive Feedback Cancellation 알고리즘 구현)

  • Jeon, Shin-Hyuk;Ji, You-Na;Park, Young-Cheol
    • The Journal of Korea Institute of Information, Electronics, and Communication Technology
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    • v.10 no.1
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    • pp.102-110
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    • 2017
  • In this paper, we have implemented an real-time adaptive feedback cancellation(AFC) algorithm that can be applied to multi-channel digital hearing aid. Multichannel digital hearing aid typically use the FFT filterbank based Wide Dynamic Range Compression(WDRC) algorithm to compensate for hearing loss. The implemented real-time acoustic feedback cancellation algorithm has one integrated structure using the same FFT filter bank with WDRC, which can be beneficial in terms of computation affecting the hearing aid battery life. In addition, when the AFC fails to operate due to nonlinear input and output, the reduction gain is applied to improve robustness in practical environment. The implemented algorithm can be further improved by adding various signal processing algorithm such as speech enhancement.

A real-time acoustic echo canceller implemented on the multimedia PC (멀티미디어 PC상에 구현된 실시간 음향 반향제거기)

  • Cha, Youn-Cheul;Yoo, Jae-Ha;Youn, Dae-Hee
    • Journal of the Korean Institute of Telematics and Electronics S
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    • v.35S no.11
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    • pp.184-193
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    • 1998
  • In this paper, a real-time acoustic echo canceller is implemented using only PC's CPU without extra help from a DSP chip. The adaptive digital filter is designed efficiently so that it can be implemented in real-time and has a proper cancellation performance. It is proposed that a new double talk detector consumes a small computational complexity and guarantees the fast detection and robust operation. The real-time acoustic echo canceller consists of the full-duplex sound card and 166 MHz Pentium PC, and requires less than 10% CPU time.

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