• 한국음향학회:학술대회논문집
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• 한국음향학회 1994년도 FIFTH WESTERN PACIFIC REGIONAL ACOUSTICS CONFERENCE SEOUL KOREA
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• pp.674-679
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• 1994

Acoustic signal such as speech and scattered sound, are generally a nonstationary process whose frequency contents vary at any instant of time. For time-varying signal, whether a nonstationary or a deterministic transient signal, a traditional frequency domain representation does not reveal the contents of signal characteristics and may lead to erroneous results such as the loss of desired characteristics features or the mis-interpretation for a wrong conclusion. A time-frequency domain representation is needed to characterize such signatures. Pseudo Wigner-Ville distribution (PWVD) is ideally suited for portraying nonstationary signal time-frequency domain and carried out by adapting the fast Fourier transform algorithm. In this paper, the important properties of PWVD were investigated using both stationary and nonstationry signatures by numerical examples PWVD was applied to acoustic sigtnatures to demonstrate its application for time-ferquency domain analysis.

• 말소리와 음성과학
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• 제6권3호
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• pp.63-72
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• 2014

The current study assessed the utility of acoustic analyses the most commonly used in routine clinical voice assessment including perturbation, nonlinear dynamic analysis, and Spectral/Cepstrum analysis based on signal typing of dysphonic voices and investigated their applicability of clinical acoustic analysis methods. A total of 70 dysphonic voice samples were classified with signal typing using narrowband spectrogram. Traditional parameters of %jitter, %shimmer, and signal-to-noise ratio were calculated for the signals using TF32 and correlation dimension(D2) of nonlinear dynamic parameter and spectral/cepstral measures including mean CPP, CPP_sd, CPPf0, CPPf0_sd, L/H ratio, and L/H ratio_sd were also calculated with ADSV(Analysis of Dysphonia in Speech and VoiceTM). Auditory perceptual analysis was performed by two blinded speech-language pathologists with GRBAS. The results showed that nearly periodic Type 1 signals were all functional dysphonia and Type 4 signals were comprised of neurogenic and organic voice disorders. Only Type 1 voice signals were reliable for perturbation analysis in this study. Significant signal typing-related differences were found in all acoustic and auditory-perceptual measures. SNR, CPP, L/H ratio values for Type 4 were significantly lower than those of other voice signals and significant higher %jitter, %shimmer were observed in Type 4 voice signals(p<.001). Additionally, with increase of signal type, D2 values significantly increased and more complex and nonlinear patterns were represented. Nevertheless, voice signals with highly noise component associated with breathiness were not able to obtain D2. In particular, CPP, was highly sensitive with voice quality 'G', 'R', 'B' than any other acoustic measures. Thus, Spectral and cepstral analyses may be applied for more severe dysphonic voices such as Type 4 signals and CPP can be more accurate and predictive acoustic marker in measuring voice quality and severity in dysphonia.

• 융합신호처리학회논문지
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• 제14권1호
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• pp.1-5
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• 2013

The two major problems related to the blood flow in replaced prosthetic heart valve are thrombus formation and hemolysis. Reliability of prosthetic valve is very important because its failure means the death of patient. There are many factors affecting the valvular failures and their representatives are mechanical failure and thrombosis, so early noninvasive detection is essentially required. The purpose of this study is to detect the various thromboses formation by using acoustic signal acquisition and its spectral analysis on the frequency domain. We made the thrombosis models using Polydimethylsiloxane (PDMS) and they are thrombosis model on the disc, around the sewing ring and fibrous tissue growth across the orifice of valve. Using microphone and amplifier, we measured the acoustic signal from the prosthetic valve, which is attached to the pulsatile mock circulation system. A/D converter sampled the acoustic signal and the spectral analysis is the main algorithm for obtaining spectrum. Then the spectrum of normal and 5 different kinds of abnormal valve were obtained. Each spectrum waveform shows a primary and secondary peak. The secondary peak changes according to the thrombus model. To quantitatively distinguish the frequency peak of the normal valve from that of the thrombosed valves, analysis using a neural network was employed. Acoustic measurement has been used as a noninvasive diagnostic tool and is thought to be a good method for detecting possible mechanical failure or thrombus.

• 한국콘텐츠학회논문지
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• 제4권2호
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• pp.1-6
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• 2004

본 논문에서는 음란 유해 사이트에서 성행위 묘사시 발생하는 음향 신호를 처리하고 분석하여 표준 패턴 음향신호와 어느 정도 일치하는지 상관 계수를 계산함으로써 내용에 기반하여 음란콘텐츠를 차단하는 방법에 대해 제안하고자 한다. 기존의 음란 유해 사이트 차단 방법은 목록기반과 단어기반에 의한 방법이어서 새로이 생겨나는 음란사이트가 차단이 안되거나 음란사이트 운영자가 교묘히 음란 단어를 변경함으로써 단어 기반으로 음란사이트가 차단이 안되는 문제가 존재했었다. 이를 해결하기 위해 본 논문에서는 콘텐츠기반의 차단법을 다루고자 한다. 특히 본 논문은 음란물의 콘텐츠(내용)에 기반한 전체 시스템중 음향 신호처리에 기반 한 음란 사이트 차단법에 대해 다루고자 한다. 끝으로 실험에 의해 제안한 방법의 유용성을 입증하고자 한다.

• 로봇학회논문지
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• 제7권4호
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• pp.231-242
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• 2012

This paper proposes particle filter(PF) method using acoustic signal for localization of an underwater robot. The method uses time of arrival(TOA) or time difference of arrival(TDOA) of acoustic signals from beacons whose locations are known. An experiment in towing tank uses TOA information. Simulation uses TDOA information and it reveals dependency of the localization performance on the uncertainty of robot motion and senor data. Also, comparison of the PF method with the least squares method of spherical interpolation(SI) and spherical intersection(SX) is provided. Since PF uses TOA or TDOA which comes from measurement of external information as well as internal motion information, its estimation is more accurate and robust to the sensor and motion uncertainty than the least squares methods.

• 한국음향학회지
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• 제22권7호
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• pp.573-578
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• 2003

최근 원격 회의 시스템에서 마이크로폰 어레이를 이용한 음원 위치 추정 및 음질 향상 기술이 연구되고 있다. 이러한 기술 가운데 대표적인 정합 필터 마이크로폰 어레이는 다수의 마이크로폰을 이용하여 음원의 위치를 추적하고 음원의 방향으로부터 입사되는 신호만을 수신할 수 있도록 한다. 그러나 이는 음향 공간의 기하학적인 형태 및 재질 등을 이용한 모델로부터 미리 계산된 음향 채널을 사용하기 때문에 실제 실내 환경과의 불일치로 인하여 성능이 저하된다. 본 논문에서는 수신된 음향 신호로부터 실내 음향 채널을 추정하고 이를 정합 필터 마이크로폰 어레이에 적용하였다 연구된 방법은 실험을 통해 그 성능을 고찰하였다.

• The Journal of the Acoustical Society of Korea
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• 제18권2E호
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• pp.45-52
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• 1999

This paper presents the integrated speech enhancement system for hands-free mobile communication. The proposed integrated system incorporates both acoustic echo cancellation and engine noise reduction device to provide signal enhancement of desired speech signal from the echoed plus noisy environments. To implement the system, a delayless subband adaptive structure is used for acoustic echo cancellation operation. The NLMS based adaptive noise canceller then applied to the residual echo removed noisy signal to achieve the selective engine noise attenuation in dominant frequency component. Two sets of computer simulations are conducted to demonstrate the effectiveness of the system; one for the fixed acoustical environment condition, the other for the robustness of the system in which, more realistic situation, the acoustic transmission environment change. Simulation results confirm the system performance of 20-25dB ERLE in acoustic echo cancellation and 9-19 dB engine noise attenuation in dominant frequency component for both cases.

• 한국음향학회지
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• 제22권5호
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• pp.351-359
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• 2003

본 논문에서는 독립성분분석을 이용한 음향 반향제거 방법을 제안하였다. 음향반향제거기의 마이크로폰에 반향 이외의 잡음이 부가될 경우 반향제거기의 성능은 저하된다. 이러한 문제를 해결하기 위해 본 연구에서는 두 개의 마이크로폰을 이용하여 반향과 선형으로 섞인 잡음을 받은 후 독립성분 분석 기법을 통해 반향과 잡음을 분리하였다. 그리고 분리된 반향 신호를 반향제거기에 사용되는 적응 알고리듬의 기준 신호로 이용함으로서 반향제거기의 성능을 향상시켰다. 컴퓨터 모의실험을 통해 제안한 방법의 타당성을 확인하였다.

• 한국전기전자재료학회논문지
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• 제25권2호
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• pp.147-152
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• 2012

This paper dealt with the frequency component analysis of acoustic signals produced by corona and series-arc discharges as a diagnostic technique for closed-switchboards. Corona and series-arc discharge were simulated by a needle-plane electrode and an arc generator specified in UL1699, respectively. Acoustic signal was detected by a wideband acoustic sensor with a frequency bandwidth of 4 Hz~100 kHz (-3 dB). We analyzed frequency spectrums of the acoustic signals detected in various discharge conditions. The results showed that acoustic signals mainly exist in ranges from 30 kHz to 60 kHz. From the experimental results, an acoustic detection system which consists of a constant current power supply (CCP), a low noise amplifier (LNA) and a band pass filter was designed and fabricated. The CCP separates the signal component from the DC source of acoustic sensor, and the LNA has a gain of 40 dB in ranges of 280 Hz~320 kHz. The high and the low cut-off frequency are 30 kHz and 60 kHz, respectively. We could detect corona and series-arc discharges without any interference by the acoustic detection system, and the best frequency is considered in ranges of 30 kHz~60 kHz.

• 한국음향학회지
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• 제20권1호
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• pp.68-76
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• 2001

유한요소법을 이용하여 압전 수중음향센서의 모델링 및 음향특성을 해석하였다. 압전 수중음향센서의 해석에서 기본적인 압전-탄성 구조물과 유체-구조물의 연성해석을 위한 유한요소 정식화를 하였으며 무한영역의 음향유체를 처리하기 위하여 IWEE (Infinite Wave Envelop Element)를 도입하였다. Tonpilz형 수중음향센서를 수중 산란체로 볼 경우 입사파가 산란체의 표면을 가진할 때 산란체로부터 발생되는 산란파는 IWEE로 인하여 무한 유체영역에서의 산란파의 감소특성을 갖게 되어 무한영역을 유한영역으로 나눈 인위적인 경계에서 반사가 일어나지 않게 되므로 산란파의 음압을 정확히 구할 수 있었다. 또한, 이러한 산란해석을 바탕으로 입사파에 대한 음향센서 내부의 전기적 응답특성인 RVS (Receiving Voltage Signal)를 구하였다. 이러한 일련의 연구 과정들은 소나 시스템을 정확히 해석하고 음향특성을 예측하는데 큰 도움이 될 것이다.