• Title/Summary/Keyword: Acoustic Signal Recognition

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Acoustic Event Detection and Matlab/Simulink Interoperation for Individualized Things-Human Interaction (사물-사람 간 개인화된 상호작용을 위한 음향신호 이벤트 감지 및 Matlab/Simulink 연동환경)

  • Lee, Sanghyun;Kim, Tag Gon;Cho, Jeonghun;Park, Daejin
    • IEMEK Journal of Embedded Systems and Applications
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    • v.10 no.4
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    • pp.189-198
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    • 2015
  • Most IoT-related approaches have tried to establish the relation by connecting the network between things. The proposed research will present how the pervasive interaction of eco-system formed by touching the objects between humans and things can be recognized on purpose. By collecting and sharing the detected patterns among all kinds of things, we can construct the environment which enables individualized interactions of different objects. To perform the aforementioned, we are going to utilize technical procedures such as event-driven signal processing, pattern matching for signal recognition, and hardware in the loop simulation. We will also aim to implement the prototype of sensor processor based on Arduino MCU, which can be integrated with system using Arduino-Matlab/Simulink hybrid-interoperation environment. In the experiment, we use piezo transducer to detect the vibration or vibrates the surface using acoustic wave, which has specific frequency spectrum and individualized signal shape in terms of time axis. The signal distortion in time and frequency domain is recorded into memory tracer within sensor processor to extract the meaningful pattern by comparing the stored with lookup table(LUT). In this paper, we will contribute the initial prototypes for the acoustic touch processor by using off-the-shelf MCU and the integrated framework based on Matlab/Simulink model to provide the individualization of the touch-sensing for the user on purpose.

An analysis of a statistical difference of acoustic Parameters' distribution between normal voice and pathological voice (병적 음성과 정상 음성의 음향학적 파라미터 분포에 대한 통계적 분석)

  • 김용주;권순복;김기련;신민철;조철우;왕수건
    • Proceedings of the IEEK Conference
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    • 2001.06d
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    • pp.249-252
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    • 2001
  • The most basic means of communication among humans is a voice. Without speaking of voice technologies, we found it is important and convenient to use a voice in everyday life. But. in consideration to speech recognition systems, we can't always desire a normal voice input as input signal to the system. Generally speaking. a pathological voice as against a normal which is a voice with a problem in the larynx. could be also special case of input voice. Of course, but the distortion of a speech signal by environmental effects i.e., noise or transmission channel was a raised problem. we will take up a pathological voices with laryngeal disease which is essential distortion factor in voice. Also, we are to find out the difference of acoustic parameters distribution between normal and pathological voice by a statistical method in our research.

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A Study on Performance Improvement Method for the Multi-Model Speech Recognition System in the DSR Environment (DSR 환경에서의 다 모델 음성 인식시스템의 성능 향상 방법에 관한 연구)

  • Jang, Hyun-Baek;Chung, Yong-Joo
    • Journal of the Institute of Convergence Signal Processing
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    • v.11 no.2
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    • pp.137-142
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    • 2010
  • Although multi-model speech recognizer has been shown to be quite successful in noisy speech recognition, the results were based on general speech front-ends which do not take into account noise adaptation techniques. In this paper, for the accurate evaluation of the multi-model based speech recognizer, we adopted a quite noise-robust speech front-end, AFE, which was proposed by the ETSI for the noisy DSR environment. For the performance comparison, the MTR which is known to give good results in the DSR environment has been used. Also, we modified the structure of the multi-model based speech recognizer to improve the recognition performance. N reference HMMs which are most similar to the input noisy speech are used as the acoustic models for recognition to cope with the errors in the selection of the reference HMMs and the noise signal variability. In addition, multiple SNR levels are used to train each of the reference HMMs to improve the robustness of the acoustic models. From the experimental results on the Aurora 2 databases, we could see better recognition rates using the modified multi-model based speech recognizer compared with the previous method.

Speech Recognition based on Environment Adaptation using SNR Mapping (SNR 매핑을 이용한 환경적응 기반 음성인식)

  • Chung, Yong-Joo
    • The Journal of the Korea institute of electronic communication sciences
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    • v.9 no.5
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    • pp.543-548
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    • 2014
  • Multiple-model based speech recognition framework (MMSR) has been known to be very successful in speech recognition. Since it uses multiple hidden Markov modes (HMMs) that corresponds to various noise types and signal-to-noise ratio (SNR) values, the selected acoustic model can have a close match with the test noisy speech. However, since the number of HMM sets is limited in practical use, the acoustic mismatch still remains as a problem. In this study, we experimentally determined the optimal SNR mapping between the test noisy speech and the HMM set to mitigate the mismatch between them. Improved performance was obtained by employing the SNR mapping instead of using the estimated SNR from the test noisy speech. When we applied the proposed method to the MMSR, the experimental results on the Aurora 2 database show that the relative word error rate reduction of 6.3% and 9.4% was achieved compared to a conventional MMSR and multi-condition training (MTR), respectively.

A Study on Speech Recognition using Vocal Tract Area Function (성도 면적 함수를 이용한 음성 인식에 관한 연구)

  • 송제혁;김동준
    • Journal of Biomedical Engineering Research
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    • v.16 no.3
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    • pp.345-352
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    • 1995
  • The LPC cepstrum coefficients, which are an acoustic features of speech signal, have been widely used as the feature parameter for various speech recognition systems and showed good performance. The vocal tract area function is a kind of articulatory feature, which is related with the physiological mechanism of speech production. This paper proposes the vocal tract area function as an alternative feature parameter for speech recognition. The linear predictive analysis using Burg algorithm and the vector quantization are performed. Then, recognition experiments for 5 Korean vowels and 10 digits are executed using the conventional LPC cepstrum coefficients and the vocal tract area function. The recognitions using the area function showed the slightly better results than those using the conventional LPC cepstrum coefficients.

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Analyzing the Acoustic Elements and Emotion Recognition from Speech Signal Based on DRNN (음향적 요소분석과 DRNN을 이용한 음성신호의 감성 인식)

  • Sim, Kwee-Bo;Park, Chang-Hyun;Joo, Young-Hoon
    • Journal of the Korean Institute of Intelligent Systems
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    • v.13 no.1
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    • pp.45-50
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    • 2003
  • Recently, robots technique has been developed remarkably. Emotion recognition is necessary to make an intimate robot. This paper shows the simulator and simulation result which recognize or classify emotions by learning pitch pattern. Also, because the pitch is not sufficient for recognizing emotion, we added acoustic elements. For that reason, we analyze the relation between emotion and acoustic elements. The simulator is composed of the DRNN(Dynamic Recurrent Neural Network), Feature extraction. DRNN is a learning algorithm for pitch pattern.

Calculus of the defect severity with EMATs by analysing the attenuation curves of the guided waves

  • Gomez, Carlos Q.;Garcia, Fausto P.;Arcos, Alfredo;Cheng, Liang;Kogia, Maria;Papelias, Mayorkinos
    • Smart Structures and Systems
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    • v.19 no.2
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    • pp.195-202
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    • 2017
  • The aim of this paper is to develop a novel method to determine the severity of a damage in a thin plate. This paper presents a novel fault detection and diagnosis approach employing a new electromagnetic acoustic transducer, called EMAT, together with a complex signal processing method. The method consists in the recognition of a fault that exists within the structure, the fault location, i.e. the identification of the geometric position of damage, and the determining the significance of the damage, which indicates the importance or severity of the defect. The main scientific novelties presented in this paper is: to develop of a new type of electromagnetic acoustic transducer; to incorporate wavelet transforms for signal representation enhancements; to investigate multi-parametric analysis for noise identification and defect classification; to study attenuation curves properties for defect localization improvement; flaw sizing and location algorithm development.

A Development of Acoustic Release System in the Seafloor (심해저용 원격 착탈 제어 시스템의 개발)

  • Kim, Young-Jin;Huh, Kyung-Moo;Jeong, Han-Cheol
    • Journal of Institute of Control, Robotics and Systems
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    • v.11 no.9
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    • pp.774-780
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    • 2005
  • For the accurate inspection of the resources and space in the ocean, the method of locating the measurement equipments in the seafloor and retrieving these equipments later after a certain period of time. is generally used. In this method, the reliability of retrieving measurement equipments is very important. In our proposed remotely-controlled acoustic release system, an underwater ultrasonic wave recognition algorithm by which we can recognize the sound signal without the influence of disturbances due to underwater environment changes is developed, and a battery is used for the reduction of electric power consumption. we show the effectiveness of our proposed system through experimental results.

Heart Sound Recognition by Analysis of wavelet transform and Neural network.

  • Lee, Jung-Jun;Lee, Sang-Min;Hong, Seung-Hong
    • Proceedings of the IEEK Conference
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    • 2000.07b
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    • pp.1045-1048
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    • 2000
  • This paper presents the application of the wavelet transform analysis and the neural network method to the phonocardiogram (PCG) signal. Heart sound is a acoustic signal generated by cardiac valves, myocardium and blood flow and is a very complex and nonstationary signal composed of many source. Heart sound can be discriminated normal heart sound and heart murmur. Murmurs have broader frequency bandwidth than the normal ones and can occur at random position of cardiac cycle. In this paper, we classified the group of heart sound as normal heart sound(NO), pre-systolic murmur(PS), early systolic murmur(ES), late systolic murmur(LS), early diastolic murmur(ED). And we used the wavelet transform to shorten artifacts and strengthen the low level signal. The ANN system was trained and tested with the back- propagation algorithm from a large data set of examples-normal and abnormal signals classified by expert. The best ANN configuration occurred with 15 hidden layer neurons. We can get the accuracy of 85.6% by using the proposed algorithm.

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Simulator for Active Sonar Target Recognition (능동소나 표적인식을 위한 시뮬레이터)

  • Seok, Jongwon;Kim, Taehwan;Bae, Keunsung
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.16 no.10
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    • pp.2137-2142
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    • 2012
  • Many studies in detection and classification of the targets in the underwater environments have been conducted for military purposes, as well as for non-military purpose. Due to the complicated characteristics of underwater acoustic signal reflecting multipath environments and spatio-temporal varying characteristics, active sonar target classification technique has been considered as a difficult technique. And it has a difficult in collecting actual underwater data. In this paper, we implemented the simulator to synthesize the active target signal, to extract feature and to classify the target in the underwater environment. In target signal synthesis, highlight and three-dimensional model are used and multi-aspect based hidden markov model is used for target classification.